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webrtc_m130/modules/audio_coding/BUILD.gn

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GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../../webrtc.gni")
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
import("audio_coding.gni")
import("//build/config/arm.gni")
Update gn files to support Mozilla build Bug: webrtc:8670 No-Presubmit: true Change-Id: I085dc63daa8274b5068540cbf56b6330f40643fa Reviewed-on: https://webrtc-review.googlesource.com/38920 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21624}
2018-01-15 10:20:00 -05:00
if (!build_with_mozilla) {
import("//third_party/protobuf/proto_library.gni")
}
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility = [ ":*" ]
Declare that rent_a_codec depends on the audio codecs That these declarations were missing was a bug, which apparently didn't actually cause build problems in either Chromium or WebRTC standalone. (Presumably, because rent_a_codec was always linked together with other build targets that did declare such dependencies.) BUG=webrtc:5435 Review URL: https://codereview.webrtc.org/1607463002 Cr-Commit-Position: refs/heads/master@{#11303}
2016-01-19 05:54:28 -08:00
audio_codec_deps = [
":cng",
":g711",
":pcm16b",
]
if (rtc_include_ilbc) {
audio_codec_deps += [ ":ilbc" ]
}
if (rtc_include_opus) {
audio_codec_deps += [ ":webrtc_opus" ]
}
Don't select audio codecs depending on GN vars `build_with_{chromium|mozilla}` BUG=webrtc:8343 Change-Id: I5943006a4da17f72eb88eae9d7ea57574d54f680 Reviewed-on: https://webrtc-review.googlesource.com/9401 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20540}
2017-11-01 15:08:12 +01:00
if (current_cpu == "arm") {
audio_codec_deps += [ ":isac_fix" ]
} else {
audio_codec_deps += [ ":isac" ]
Declare that rent_a_codec depends on the audio codecs That these declarations were missing was a bug, which apparently didn't actually cause build problems in either Chromium or WebRTC standalone. (Presumably, because rent_a_codec was always linked together with other build targets that did declare such dependencies.) BUG=webrtc:5435 Review URL: https://codereview.webrtc.org/1607463002 Cr-Commit-Position: refs/heads/master@{#11303}
2016-01-19 05:54:28 -08:00
}
Don't select audio codecs depending on GN vars `build_with_{chromium|mozilla}` BUG=webrtc:8343 Change-Id: I5943006a4da17f72eb88eae9d7ea57574d54f680 Reviewed-on: https://webrtc-review.googlesource.com/9401 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20540}
2017-11-01 15:08:12 +01:00
audio_codec_deps += [ ":g722" ]
Declare that rent_a_codec depends on the audio codecs That these declarations were missing was a bug, which apparently didn't actually cause build problems in either Chromium or WebRTC standalone. (Presumably, because rent_a_codec was always linked together with other build targets that did declare such dependencies.) BUG=webrtc:5435 Review URL: https://codereview.webrtc.org/1607463002 Cr-Commit-Position: refs/heads/master@{#11303}
2016-01-19 05:54:28 -08:00
if (!build_with_mozilla && !build_with_chromium) {
audio_codec_deps += [ ":red" ]
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
audio_coding_deps = audio_codec_deps + [
"../..:webrtc_common",
"../../common_audio",
"../../system_wrappers",
]
Declare that rent_a_codec depends on the audio codecs That these declarations were missing was a bug, which apparently didn't actually cause build problems in either Chromium or WebRTC standalone. (Presumably, because rent_a_codec was always linked together with other build targets that did declare such dependencies.) BUG=webrtc:5435 Review URL: https://codereview.webrtc.org/1607463002 Cr-Commit-Position: refs/heads/master@{#11303}
2016-01-19 05:54:28 -08:00
Stop using old AudioCodingModule::RegisterReceiveCodec overloads BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/2388153004 Cr-Commit-Position: refs/heads/master@{#14753}
2016-10-24 13:47:09 -07:00
rtc_static_library("audio_format_conversion") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Stop using old AudioCodingModule::RegisterReceiveCodec overloads BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/2388153004 Cr-Commit-Position: refs/heads/master@{#14753}
2016-10-24 13:47:09 -07:00
sources = [
"codecs/audio_format_conversion.cc",
"codecs/audio_format_conversion.h",
]
deps = [
Use relative names in GN to make Chromium happy A recent CL (https://codereview.chromium.org/2388153004/) introduced absolute names, which caused Chromium builds to fail. TBR=kjellander@webrtc.org BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/2446643005 Cr-Commit-Position: refs/heads/master@{#14755}
2016-10-24 16:31:17 -07:00
"../..:webrtc_common",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:sanitizer",
Stop using old AudioCodingModule::RegisterReceiveCodec overloads BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/2388153004 Cr-Commit-Position: refs/heads/master@{#14753}
2016-10-24 13:47:09 -07:00
]
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("rent_a_codec") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
RentACodec: New class that takes over part of ACMCodecDB's job Following CLs will finish the takeover completely. After that, RentACodec will also start creating and owning codecs, at which point its name will start making sense. BUG=webrtc:5028 Review URL: https://codereview.webrtc.org/1412683006 Cr-Commit-Position: refs/heads/master@{#10432}
2015-10-27 11:40:24 -07:00
sources = [
audio_coding: remove "main" directory This is the last piece of the old directory layout of the modules. Duplicated header files are left in audio_coding/main/include until downstream code is updated to the new location. They have pragma warnings added to them and identical header guards as the new headers to avoid breaking things. BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1481493004 Cr-Commit-Position: refs/heads/master@{#10803}
2015-11-26 04:44:54 -08:00
"acm2/acm_codec_database.cc",
"acm2/acm_codec_database.h",
"acm2/rent_a_codec.cc",
"acm2/rent_a_codec.h",
RentACodec: New class that takes over part of ACMCodecDB's job Following CLs will finish the takeover completely. After that, RentACodec will also start creating and owning codecs, at which point its name will start making sense. BUG=webrtc:5028 Review URL: https://codereview.webrtc.org/1412683006 Cr-Commit-Position: refs/heads/master@{#10432}
2015-10-27 11:40:24 -07:00
]
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
deps = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
"../../system_wrappers",
":audio_coding_module_typedefs",
":isac_common",
":isac_fix_c",
":neteq_decoder_enum",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
] + audio_codec_deps
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ ) Reason for revert: Trying to re-land after solving some related issues. There are no changes compared to the original CL. Original issue's description: > Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ ) > > Reason for revert: > I will try to reland next week because it is causing some problems. > > Original issue's description: > > To accommodate some downstream WebRTC users we need to loosen > > the coupling between our code and the //third_party/protobuf. > > > > This includes using typedefs to define strings instead of > > assuming std::string. > > > > After this refactoring it will be possible to link with other > > protobuf implementations than the current one. > > > > We moved the PRESUBMIT check to another CL [1]. The goal of this > > presubmit is to avoid the direct usage of google::protobuf outside > > of the webrtc/base/protobuf_utils.h header file. > > > > [1] - https://codereview.webrtc.org/2753823003/ > > > > BUG=webrtc:7340 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2747863003 > > Cr-Commit-Position: refs/heads/master@{#17466} > > Committed: https://chromium.googlesource.com/external/webrtc/+/16ab93b952f9e8268f2e663ffe49548e8043d5af > > TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:7340 > > Review-Url: https://codereview.webrtc.org/2786363002 > Cr-Commit-Position: refs/heads/master@{#17483} > Committed: https://chromium.googlesource.com/external/webrtc/+/d00aad5eb2fa5a7b5aeda714f7702b50cd26ee28 TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7340 NOTRY=True Review-Url: https://codereview.webrtc.org/2791963003 Cr-Commit-Position: refs/heads/master@{#17584}
2017-04-07 00:59:12 -07:00
Declare that rent_a_codec depends on the audio codecs That these declarations were missing was a bug, which apparently didn't actually cause build problems in either Chromium or WebRTC standalone. (Presumably, because rent_a_codec was always linked together with other build targets that did declare such dependencies.) BUG=webrtc:5435 Review URL: https://codereview.webrtc.org/1607463002 Cr-Commit-Position: refs/heads/master@{#11303}
2016-01-19 05:54:28 -08:00
defines = audio_codec_defines
RentACodec: New class that takes over part of ACMCodecDB's job Following CLs will finish the takeover completely. After that, RentACodec will also start creating and owning codecs, at which point its name will start making sense. BUG=webrtc:5028 Review URL: https://codereview.webrtc.org/1412683006 Cr-Commit-Position: refs/heads/master@{#10432}
2015-10-27 11:40:24 -07:00
}
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
config("audio_coding_config") {
include_dirs = [
audio_coding: remove "main" directory This is the last piece of the old directory layout of the modules. Duplicated header files are left in audio_coding/main/include until downstream code is updated to the new location. They have pragma warnings added to them and identical header guards as the new headers to avoid breaking things. BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1481493004 Cr-Commit-Position: refs/heads/master@{#10803}
2015-11-26 04:44:54 -08:00
"include",
modules: more interface -> include renames This changes the following module directories: * webrtc/modules/audio_conference_mixer/interface * webrtc/modules/interface * webrtc/modules/media_file/interface * webrtc/modules/rtp_rtcp/interface * webrtc/modules/utility/interface To avoid breaking downstream, I followed this recipe: 1. Copy the interface dir to a new sibling directory: include 2. Update the header guards in the include directory to match the style guide. 3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code. 4. Add a pragma warning in the header files in the interface dir. Example: #pragma message("WARNING: webrtc/modules/interface is DEPRECATED; " "use webrtc/modules/include") 5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S) 6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*) BUG=5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417683006 . Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 08:31:52 +01:00
"../include",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
}
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
rtc_source_set("audio_coding_module_typedefs") {
sources = [
"include/audio_coding_module_typedefs.h",
]
deps = [
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
"../..:webrtc_common",
]
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("audio_coding") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
audio_coding: remove "main" directory This is the last piece of the old directory layout of the modules. Duplicated header files are left in audio_coding/main/include until downstream code is updated to the new location. They have pragma warnings added to them and identical header guards as the new headers to avoid breaking things. BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1481493004 Cr-Commit-Position: refs/heads/master@{#10803}
2015-11-26 04:44:54 -08:00
"acm2/acm_receiver.cc",
"acm2/acm_receiver.h",
"acm2/acm_resampler.cc",
"acm2/acm_resampler.h",
"acm2/audio_coding_module.cc",
"acm2/call_statistics.cc",
"acm2/call_statistics.h",
"acm2/codec_manager.cc",
"acm2/codec_manager.h",
"include/audio_coding_module.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
defines = []
GN Templates: Move common_inherited_config to the template. Remove common_inherited_config from the targets and add it to the template instead. BUG=webrtc:6187 NOTRY=True Review-Url: https://codereview.webrtc.org/2311843002 Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 06:10:18 -07:00
public_configs = [ ":audio_coding_config" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (rtc_include_opus) {
public_deps = [
":webrtc_opus",
]
}
GN: Fix Windows build. This required a tiny include fix in src/third_party/winsdk_samples/src which was committed in https://code.google.com/p/webrtc/source/detail?r=7951 This incorporates contribution from vchigrin@yandex-team.ru in https://webrtc-codereview.appspot.com/29299004/ BUG=261,1348,4105 R=pbos@webrtc.org TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8027 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 21:22:01 +00:00
if (is_win) {
cflags = [
# TODO(kjellander): Bug 261: fix this warning.
"/wd4373", # virtual function override.
]
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
deps = audio_coding_deps + [
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../../rtc_base:deprecation",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:metrics_api",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
":audio_coding_module_typedefs",
Declare that rent_a_codec depends on the audio codecs That these declarations were missing was a bug, which apparently didn't actually cause build problems in either Chromium or WebRTC standalone. (Presumably, because rent_a_codec was always linked together with other build targets that did declare such dependencies.) BUG=webrtc:5435 Review URL: https://codereview.webrtc.org/1607463002 Cr-Commit-Position: refs/heads/master@{#11303}
2016-01-19 05:54:28 -08:00
":neteq",
":rent_a_codec",
This CL removes all usages of our custom ostream << overloads. This prepares us for removing them altogether. Bug: webrtc:8982 Change-Id: I66002cc8d4bf0e07925766d568d2498422f0f38e Reviewed-on: https://webrtc-review.googlesource.com/64142 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Henrik Grunell <henrikg@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22707}
2018-04-03 13:40:05 +02:00
"../../rtc_base:audio_format_to_string",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Moved RtcEventLog files from call/ to logging/ The RtcEventLog headers need to be accessible from any place which needs logging, and the implementation needs access to data structures that are logged. After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future). The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/). This change allows using RtcEventLog in the p2p/ directory, so that we can log STUN pings and ICE state transitions. BUG=webrtc:6393 R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org Review URL: https://codereview.webrtc.org/2380683005 . Cr-Commit-Position: refs/heads/master@{#14485}
2016-10-03 18:31:22 -07:00
"../../logging:rtc_event_log_api",
Declare that rent_a_codec depends on the audio codecs That these declarations were missing was a bug, which apparently didn't actually cause build problems in either Chromium or WebRTC standalone. (Presumably, because rent_a_codec was always linked together with other build targets that did declare such dependencies.) BUG=webrtc:5435 Review URL: https://codereview.webrtc.org/1607463002 Cr-Commit-Position: refs/heads/master@{#11303}
2016-01-19 05:54:28 -08:00
]
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
defines = audio_coding_defines
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
rtc_static_library("legacy_encoded_audio_frame") {
Move the AudioDecoder interface out of NetEq It belongs with the codecs, next to the AudioEncoder interface. R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798 and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799 Review URL: https://webrtc-codereview.appspot.com/27309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:12:53 +00:00
sources = [
Moved legacy_encoded_audio_frame into audio_decoder_interface. audio_decoder.cc depends on LegacyEncodedAudioFrame and LegacyEncodedAudioFrame depends on AudioDecoder::EncodedAudioFrame, so there's no clear way to separate them as of now. This error is also hodling up builds downstream. I expect we'll revisit these dependencies as part of the upcoming larger restructuring effort. NOTRY=true BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2359763002 Cr-Commit-Position: refs/heads/master@{#14329}
2016-09-21 05:50:37 -07:00
"codecs/legacy_encoded_audio_frame.cc",
"codecs/legacy_encoded_audio_frame.h",
Move the AudioDecoder interface out of NetEq It belongs with the codecs, next to the AudioEncoder interface. R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798 and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799 Review URL: https://webrtc-codereview.appspot.com/27309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:12:53 +00:00
]
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
Move the AudioDecoder interface out of NetEq It belongs with the codecs, next to the AudioEncoder interface. R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798 and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799 Review URL: https://webrtc-codereview.appspot.com/27309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:12:53 +00:00
}
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
config("cng_config") {
Removing useless include_dirs entry. After the migration from serc/webrtc to src/ this entry in the include_dirs list is not needed anymore. Bug: chromium:611808 Change-Id: I17c87509b73b8a44f758d59ada28d366da664649 Reviewed-on: https://webrtc-review.googlesource.com/1920 Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19894}
2017-09-18 13:09:16 +02:00
include_dirs = [ "codecs/cng/include" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("cng") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
Make an AudioEncoder subclass for comfort noise BUG=3926 R=bjornv@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7857 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 07:29:08 +00:00
"codecs/cng/audio_encoder_cng.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/cng/audio_encoder_cng.h",
Convert CNG into C++ and remove it from AudioDecoder Broke out CNG from AudioDecoder as they didn't really share an interface. Converted the CNG code to C++, to make initialization and resource handling easier. This includes several changes to the behavior, favoring RTC_CHECKs over returning error codes. Review URL: https://codereview.webrtc.org/1868143002 Cr-Commit-Position: refs/heads/master@{#12491}
2016-04-25 07:55:58 -07:00
"codecs/cng/webrtc_cng.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/cng/webrtc_cng.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
GN Templates: Move common_inherited_config to the template. Remove common_inherited_config from the targets and add it to the template instead. BUG=webrtc:6187 NOTRY=True Review-Url: https://codereview.webrtc.org/2311843002 Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 06:10:18 -07:00
public_configs = [ ":cng_config" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
deps = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../../api/audio_codecs:audio_codecs_api",
Make the audio codecs' GN targets self-sufficient Also running "gn format" on the file. R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1494993002 . Cr-Commit-Position: refs/heads/master@{#10886}
2015-12-03 17:47:21 +01:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
]
}
config("red_config") {
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
include_dirs = [ "codecs/red" ]
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("red") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
sources = [
"codecs/red/audio_encoder_copy_red.cc",
"codecs/red/audio_encoder_copy_red.h",
]
GN Templates: Move common_inherited_config to the template. Remove common_inherited_config from the targets and add it to the template instead. BUG=webrtc:6187 NOTRY=True Review-Url: https://codereview.webrtc.org/2311843002 Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 06:10:18 -07:00
public_configs = [ ":red_config" ]
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
deps = [
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../../api/audio_codecs:audio_codecs_api",
Make the audio codecs' GN targets self-sufficient Also running "gn format" on the file. R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1494993002 . Cr-Commit-Position: refs/heads/master@{#10886}
2015-12-03 17:47:21 +01:00
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
config("g711_config") {
Removing useless include_dirs entry. After the migration from serc/webrtc to src/ this entry in the include_dirs list is not needed anymore. Bug: chromium:611808 Change-Id: I17c87509b73b8a44f758d59ada28d366da664649 Reviewed-on: https://webrtc-review.googlesource.com/1920 Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19894}
2017-09-18 13:09:16 +02:00
include_dirs = [ "codecs/g711/include" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("g711") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
Move AudioDecoderPcm* next to AudioEncoderPcm* All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1348613003 Cr-Commit-Position: refs/heads/master@{#10015}
2015-09-22 06:16:51 -07:00
"codecs/g711/audio_decoder_pcm.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/g711/audio_decoder_pcm.h",
Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/29799004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7481 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 12:48:29 +00:00
"codecs/g711/audio_encoder_pcm.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/g711/audio_encoder_pcm.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
GN Templates: Move common_inherited_config to the template. Remove common_inherited_config from the targets and add it to the template instead. BUG=webrtc:6187 NOTRY=True Review-Url: https://codereview.webrtc.org/2311843002 Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 06:10:18 -07:00
public_configs = [ ":g711_config" ]
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
":legacy_encoded_audio_frame",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
public_deps = [
":g711_c",
]
}
rtc_source_set("g711_c") {
sources = [
"codecs/g711/g711.c",
"codecs/g711/g711.h",
"codecs/g711/g711_interface.c",
"codecs/g711/g711_interface.h",
]
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
deps = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
config("g722_config") {
Removing useless include_dirs entry. After the migration from serc/webrtc to src/ this entry in the include_dirs list is not needed anymore. Bug: chromium:611808 Change-Id: I17c87509b73b8a44f758d59ada28d366da664649 Reviewed-on: https://webrtc-review.googlesource.com/1920 Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19894}
2017-09-18 13:09:16 +02:00
include_dirs = [ "codecs/g722/include" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("g722") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
Move AudioDecoderG722 next to AudioEncoderG722 All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1346993002 Cr-Commit-Position: refs/heads/master@{#9966}
2015-09-17 03:12:35 -07:00
"codecs/g722/audio_decoder_g722.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/g722/audio_decoder_g722.h",
AudioEncoder subclass for G722 BUG=3926 R=henrik.lundin@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7779 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 11:45:51 +00:00
"codecs/g722/audio_encoder_g722.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/g722/audio_encoder_g722.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
GN Templates: Move common_inherited_config to the template. Remove common_inherited_config from the targets and add it to the template instead. BUG=webrtc:6187 NOTRY=True Review-Url: https://codereview.webrtc.org/2311843002 Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 06:10:18 -07:00
public_configs = [ ":g722_config" ]
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
":legacy_encoded_audio_frame",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
G722 implementation of the AudioEncoderFactoryTemplate API Now the templated AudioEncoderFactory can create G722 encoders! BUG=webrtc:7833 Review-Url: https://codereview.webrtc.org/2934833002 Cr-Commit-Position: refs/heads/master@{#18644}
2017-06-17 17:41:59 -07:00
"../../api/audio_codecs/g722:audio_encoder_g722_config",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
public_deps = [
":g722_c",
]
}
rtc_source_set("g722_c") {
sources = [
"codecs/g722/g722_decode.c",
"codecs/g722/g722_enc_dec.h",
"codecs/g722/g722_encode.c",
"codecs/g722/g722_interface.c",
"codecs/g722/g722_interface.h",
]
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
deps = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
config("ilbc_config") {
Removing useless include_dirs entry. After the migration from serc/webrtc to src/ this entry in the include_dirs list is not needed anymore. Bug: chromium:611808 Change-Id: I17c87509b73b8a44f758d59ada28d366da664649 Reviewed-on: https://webrtc-review.googlesource.com/1920 Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19894}
2017-09-18 13:09:16 +02:00
include_dirs = [ "codecs/ilbc/include" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("ilbc") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
Move AudioDecoderIlbc next to AudioEncoderIlbc All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1348053002 Cr-Commit-Position: refs/heads/master@{#9961}
2015-09-16 21:26:32 -07:00
"codecs/ilbc/audio_decoder_ilbc.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/ilbc/audio_decoder_ilbc.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/ilbc/audio_encoder_ilbc.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/ilbc/audio_encoder_ilbc.h",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
]
public_configs = [ ":ilbc_config" ]
deps = [
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
":legacy_encoded_audio_frame",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Expose ILBC codec in webrtc/api/audio_codecs/ BUG=webrtc:7834, webrtc:7840 Review-Url: https://codereview.webrtc.org/2951873002 Cr-Commit-Position: refs/heads/master@{#18803}
2017-06-28 02:05:04 -07:00
"../../api/audio_codecs/ilbc:audio_encoder_ilbc_config",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
]
public_deps = [
":ilbc_c",
]
}
rtc_source_set("ilbc_c") {
sources = [
"codecs/ilbc/abs_quant.c",
"codecs/ilbc/abs_quant.h",
"codecs/ilbc/abs_quant_loop.c",
"codecs/ilbc/abs_quant_loop.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/ilbc/augmented_cb_corr.c",
"codecs/ilbc/augmented_cb_corr.h",
"codecs/ilbc/bw_expand.c",
"codecs/ilbc/bw_expand.h",
"codecs/ilbc/cb_construct.c",
"codecs/ilbc/cb_construct.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/ilbc/cb_mem_energy.c",
"codecs/ilbc/cb_mem_energy.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/ilbc/cb_mem_energy_augmentation.c",
"codecs/ilbc/cb_mem_energy_augmentation.h",
"codecs/ilbc/cb_mem_energy_calc.c",
"codecs/ilbc/cb_mem_energy_calc.h",
"codecs/ilbc/cb_search.c",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/ilbc/cb_search.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/ilbc/cb_search_core.c",
"codecs/ilbc/cb_search_core.h",
"codecs/ilbc/cb_update_best_index.c",
"codecs/ilbc/cb_update_best_index.h",
"codecs/ilbc/chebyshev.c",
"codecs/ilbc/chebyshev.h",
"codecs/ilbc/comp_corr.c",
"codecs/ilbc/comp_corr.h",
"codecs/ilbc/constants.c",
"codecs/ilbc/constants.h",
"codecs/ilbc/create_augmented_vec.c",
"codecs/ilbc/create_augmented_vec.h",
"codecs/ilbc/decode.c",
"codecs/ilbc/decode.h",
"codecs/ilbc/decode_residual.c",
"codecs/ilbc/decode_residual.h",
"codecs/ilbc/decoder_interpolate_lsf.c",
"codecs/ilbc/decoder_interpolate_lsf.h",
"codecs/ilbc/defines.h",
"codecs/ilbc/do_plc.c",
"codecs/ilbc/do_plc.h",
"codecs/ilbc/encode.c",
"codecs/ilbc/encode.h",
"codecs/ilbc/energy_inverse.c",
"codecs/ilbc/energy_inverse.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/ilbc/enh_upsample.c",
"codecs/ilbc/enh_upsample.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/ilbc/enhancer.c",
"codecs/ilbc/enhancer.h",
"codecs/ilbc/enhancer_interface.c",
"codecs/ilbc/enhancer_interface.h",
"codecs/ilbc/filtered_cb_vecs.c",
"codecs/ilbc/filtered_cb_vecs.h",
"codecs/ilbc/frame_classify.c",
"codecs/ilbc/frame_classify.h",
"codecs/ilbc/gain_dequant.c",
"codecs/ilbc/gain_dequant.h",
"codecs/ilbc/gain_quant.c",
"codecs/ilbc/gain_quant.h",
"codecs/ilbc/get_cd_vec.c",
"codecs/ilbc/get_cd_vec.h",
"codecs/ilbc/get_lsp_poly.c",
"codecs/ilbc/get_lsp_poly.h",
"codecs/ilbc/get_sync_seq.c",
"codecs/ilbc/get_sync_seq.h",
"codecs/ilbc/hp_input.c",
"codecs/ilbc/hp_input.h",
"codecs/ilbc/hp_output.c",
"codecs/ilbc/hp_output.h",
"codecs/ilbc/ilbc.c",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/ilbc/ilbc.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/ilbc/index_conv_dec.c",
"codecs/ilbc/index_conv_dec.h",
"codecs/ilbc/index_conv_enc.c",
"codecs/ilbc/index_conv_enc.h",
"codecs/ilbc/init_decode.c",
"codecs/ilbc/init_decode.h",
"codecs/ilbc/init_encode.c",
"codecs/ilbc/init_encode.h",
"codecs/ilbc/interpolate.c",
"codecs/ilbc/interpolate.h",
"codecs/ilbc/interpolate_samples.c",
"codecs/ilbc/interpolate_samples.h",
"codecs/ilbc/lpc_encode.c",
"codecs/ilbc/lpc_encode.h",
"codecs/ilbc/lsf_check.c",
"codecs/ilbc/lsf_check.h",
"codecs/ilbc/lsf_interpolate_to_poly_dec.c",
"codecs/ilbc/lsf_interpolate_to_poly_dec.h",
"codecs/ilbc/lsf_interpolate_to_poly_enc.c",
"codecs/ilbc/lsf_interpolate_to_poly_enc.h",
"codecs/ilbc/lsf_to_lsp.c",
"codecs/ilbc/lsf_to_lsp.h",
"codecs/ilbc/lsf_to_poly.c",
"codecs/ilbc/lsf_to_poly.h",
"codecs/ilbc/lsp_to_lsf.c",
"codecs/ilbc/lsp_to_lsf.h",
"codecs/ilbc/my_corr.c",
"codecs/ilbc/my_corr.h",
"codecs/ilbc/nearest_neighbor.c",
"codecs/ilbc/nearest_neighbor.h",
"codecs/ilbc/pack_bits.c",
"codecs/ilbc/pack_bits.h",
"codecs/ilbc/poly_to_lsf.c",
"codecs/ilbc/poly_to_lsf.h",
"codecs/ilbc/poly_to_lsp.c",
"codecs/ilbc/poly_to_lsp.h",
"codecs/ilbc/refiner.c",
"codecs/ilbc/refiner.h",
"codecs/ilbc/simple_interpolate_lsf.c",
"codecs/ilbc/simple_interpolate_lsf.h",
"codecs/ilbc/simple_lpc_analysis.c",
"codecs/ilbc/simple_lpc_analysis.h",
"codecs/ilbc/simple_lsf_dequant.c",
"codecs/ilbc/simple_lsf_dequant.h",
"codecs/ilbc/simple_lsf_quant.c",
"codecs/ilbc/simple_lsf_quant.h",
"codecs/ilbc/smooth.c",
"codecs/ilbc/smooth.h",
"codecs/ilbc/smooth_out_data.c",
"codecs/ilbc/smooth_out_data.h",
"codecs/ilbc/sort_sq.c",
"codecs/ilbc/sort_sq.h",
"codecs/ilbc/split_vq.c",
"codecs/ilbc/split_vq.h",
"codecs/ilbc/state_construct.c",
"codecs/ilbc/state_construct.h",
"codecs/ilbc/state_search.c",
"codecs/ilbc/state_search.h",
"codecs/ilbc/swap_bytes.c",
"codecs/ilbc/swap_bytes.h",
"codecs/ilbc/unpack_bits.c",
"codecs/ilbc/unpack_bits.h",
"codecs/ilbc/vq3.c",
"codecs/ilbc/vq3.h",
"codecs/ilbc/vq4.c",
"codecs/ilbc/vq4.h",
"codecs/ilbc/window32_w32.c",
"codecs/ilbc/window32_w32.h",
"codecs/ilbc/xcorr_coef.c",
"codecs/ilbc/xcorr_coef.h",
]
GN Templates: Move common_inherited_config to the template. Remove common_inherited_config from the targets and add it to the template instead. BUG=webrtc:6187 NOTRY=True Review-Url: https://codereview.webrtc.org/2311843002 Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 06:10:18 -07:00
public_configs = [ ":ilbc_config" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
deps = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Make the audio codecs' GN targets self-sufficient Also running "gn format" on the file. R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1494993002 . Cr-Commit-Position: refs/heads/master@{#10886}
2015-12-03 17:47:21 +01:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:sanitizer",
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("isac_common") {
iSAC: Make separate AudioEncoder and AudioDecoder objects The only shared state is now the bandwidth estimation info. This reduces the amount and complexity of the locking substantially. Review URL: https://codereview.webrtc.org/1208993010 Cr-Commit-Position: refs/heads/master@{#9762}
2015-08-24 02:03:23 -07:00
sources = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"codecs/isac/audio_decoder_isac_t.h",
"codecs/isac/audio_decoder_isac_t_impl.h",
iSAC: Make separate AudioEncoder and AudioDecoder objects The only shared state is now the bandwidth estimation info. This reduces the amount and complexity of the locking substantially. Review URL: https://codereview.webrtc.org/1208993010 Cr-Commit-Position: refs/heads/master@{#9762}
2015-08-24 02:03:23 -07:00
"codecs/isac/audio_encoder_isac_t.h",
"codecs/isac/audio_encoder_isac_t_impl.h",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"codecs/isac/bandwidth_info.h",
iSAC: Make separate AudioEncoder and AudioDecoder objects The only shared state is now the bandwidth estimation info. This reduces the amount and complexity of the locking substantially. Review URL: https://codereview.webrtc.org/1208993010 Cr-Commit-Position: refs/heads/master@{#9762}
2015-08-24 02:03:23 -07:00
"codecs/isac/locked_bandwidth_info.cc",
"codecs/isac/locked_bandwidth_info.h",
]
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
deps = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
"../..:webrtc_common",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../../api:optional",
Injectable audio encoders: BuiltinAudioEncoderFactory This CL contains all the changes made to audio_coding while making audio encoders injectable. Apart from some small changes to webrtcvoiceengine, nothing here is hooked up to the outside world. Those changes will be added to a follow-up CL. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2695243005 Cr-Commit-Position: refs/heads/master@{#17569}
2017-04-06 10:03:21 -07:00
"../../api/audio_codecs:audio_codecs_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
]
iSAC: Make separate AudioEncoder and AudioDecoder objects The only shared state is now the bandwidth estimation info. This reduces the amount and complexity of the locking substantially. Review URL: https://codereview.webrtc.org/1208993010 Cr-Commit-Position: refs/heads/master@{#9762}
2015-08-24 02:03:23 -07:00
}
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
config("isac_config") {
Removing useless include_dirs entry. After the migration from serc/webrtc to src/ this entry in the include_dirs list is not needed anymore. Bug: chromium:611808 Change-Id: I17c87509b73b8a44f758d59ada28d366da664649 Reviewed-on: https://webrtc-review.googlesource.com/1920 Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19894}
2017-09-18 13:09:16 +02:00
include_dirs = [ "codecs/isac/main/include" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("isac") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
sources = [
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
"codecs/isac/main/include/audio_decoder_isac.h",
"codecs/isac/main/include/audio_encoder_isac.h",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
"codecs/isac/main/source/audio_decoder_isac.cc",
"codecs/isac/main/source/audio_encoder_isac.cc",
]
deps = [
":isac_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
]
public_deps = [
":isac_c",
]
}
rtc_static_library("isac_c") {
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
audio_coding: rename interface -> include BUG=webrtc:5095 R=henrik.lundin@webrtc.org TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417173004 . Cr-Commit-Position: refs/heads/master@{#10444}
2015-10-29 11:31:02 +01:00
"codecs/isac/main/include/isac.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/main/source/arith_routines.c",
"codecs/isac/main/source/arith_routines.h",
"codecs/isac/main/source/arith_routines_hist.c",
"codecs/isac/main/source/arith_routines_logist.c",
"codecs/isac/main/source/bandwidth_estimator.c",
"codecs/isac/main/source/bandwidth_estimator.h",
"codecs/isac/main/source/codec.h",
"codecs/isac/main/source/crc.c",
"codecs/isac/main/source/crc.h",
"codecs/isac/main/source/decode.c",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/isac/main/source/decode_bwe.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/main/source/encode.c",
"codecs/isac/main/source/encode_lpc_swb.c",
"codecs/isac/main/source/encode_lpc_swb.h",
"codecs/isac/main/source/entropy_coding.c",
"codecs/isac/main/source/entropy_coding.h",
"codecs/isac/main/source/fft.c",
"codecs/isac/main/source/fft.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/isac/main/source/filter_functions.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/main/source/filterbank_tables.c",
"codecs/isac/main/source/filterbank_tables.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/isac/main/source/filterbanks.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/main/source/intialize.c",
"codecs/isac/main/source/isac.c",
Move AudioDecoderIsac* to its own files Currently, it's sitting in AudioEncoderIsac*'s files, which is less than obvious. This CL puts the encoder and decoder in separate files together with the C implementation; CLs are afoot to make it so for the other built-in codecs as well. BUG=webrtc:4557 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1339253003 . Cr-Commit-Position: refs/heads/master@{#10018}
2015-09-22 19:31:40 +02:00
"codecs/isac/main/source/isac_float_type.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/main/source/lattice.c",
"codecs/isac/main/source/lpc_analysis.c",
"codecs/isac/main/source/lpc_analysis.h",
"codecs/isac/main/source/lpc_gain_swb_tables.c",
"codecs/isac/main/source/lpc_gain_swb_tables.h",
"codecs/isac/main/source/lpc_shape_swb12_tables.c",
"codecs/isac/main/source/lpc_shape_swb12_tables.h",
"codecs/isac/main/source/lpc_shape_swb16_tables.c",
"codecs/isac/main/source/lpc_shape_swb16_tables.h",
"codecs/isac/main/source/lpc_tables.c",
"codecs/isac/main/source/lpc_tables.h",
"codecs/isac/main/source/os_specific_inline.h",
"codecs/isac/main/source/pitch_estimator.c",
"codecs/isac/main/source/pitch_estimator.h",
"codecs/isac/main/source/pitch_filter.c",
"codecs/isac/main/source/pitch_gain_tables.c",
"codecs/isac/main/source/pitch_gain_tables.h",
"codecs/isac/main/source/pitch_lag_tables.c",
"codecs/isac/main/source/pitch_lag_tables.h",
"codecs/isac/main/source/settings.h",
"codecs/isac/main/source/spectrum_ar_model_tables.c",
"codecs/isac/main/source/spectrum_ar_model_tables.h",
"codecs/isac/main/source/structs.h",
"codecs/isac/main/source/transform.c",
]
if (is_linux) {
libs = [ "m" ]
}
GN Templates: Move common_inherited_config to the template. Remove common_inherited_config from the targets and add it to the template instead. BUG=webrtc:6187 NOTRY=True Review-Url: https://codereview.webrtc.org/2311843002 Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 06:10:18 -07:00
public_configs = [ ":isac_config" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
Make an AudioEncoder subclass for iSAC BUG=3926 Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675 and reverted: https://code.google.com/p/webrtc/source/detail?r=7676 R=henrik.lundin@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:08:19 +00:00
deps = [
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
":isac_common",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Migrated ILBC and ISAC test targets for GN. Migrated GN targets ilbc_test, isac_api_test, isac_switch_samprate_test from webrtc/modules/audio_coding/codecs NOTRY=True NOPRESUBMIT=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2270403002 Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 04:09:19 -07:00
"../..:webrtc_common",
Make an AudioEncoder subclass for iSAC BUG=3926 Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675 and reverted: https://code.google.com/p/webrtc/source/detail?r=7676 R=henrik.lundin@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:08:19 +00:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:compile_assert_c",
"../../rtc_base:rtc_base_approved",
Make an AudioEncoder subclass for iSAC BUG=3926 Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675 and reverted: https://code.google.com/p/webrtc/source/detail?r=7676 R=henrik.lundin@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:08:19 +00:00
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
config("isac_fix_config") {
Removing useless include_dirs entry. After the migration from serc/webrtc to src/ this entry in the include_dirs list is not needed anymore. Bug: chromium:611808 Change-Id: I17c87509b73b8a44f758d59ada28d366da664649 Reviewed-on: https://webrtc-review.googlesource.com/1920 Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19894}
2017-09-18 13:09:16 +02:00
include_dirs = [ "codecs/isac/fix/include" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("isac_fix") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
sources = [
"codecs/isac/fix/source/audio_decoder_isacfix.cc",
"codecs/isac/fix/source/audio_encoder_isacfix.cc",
]
public_configs = [ ":isac_fix_config" ]
deps = [
":isac_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
"../../common_audio",
"../../system_wrappers",
]
public_deps = [
":isac_fix_c",
]
if (rtc_build_with_neon) {
deps += [ ":isac_neon" ]
}
}
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
rtc_source_set("isac_fix_common") {
sources = [
"codecs/isac/fix/source/codec.h",
Using fully qualified #include paths in isac code. WebRTC internal code should always used include paths that starts from the root of the project and that clearly identify the header file. This allows 'gn check' to actually keep dependencies under control because 'gn check' cannot enforce anything if the include path is not fully qualified (starting from the root of the project). Bug: webrtc:8815 Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc Reviewed-on: https://webrtc-review.googlesource.com/46101 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21847}
2018-02-01 14:12:55 +01:00
"codecs/isac/fix/source/entropy_coding.h",
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
"codecs/isac/fix/source/fft.c",
"codecs/isac/fix/source/fft.h",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"codecs/isac/fix/source/filterbank_internal.h",
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
"codecs/isac/fix/source/settings.h",
Using fully qualified #include paths in isac code. WebRTC internal code should always used include paths that starts from the root of the project and that clearly identify the header file. This allows 'gn check' to actually keep dependencies under control because 'gn check' cannot enforce anything if the include path is not fully qualified (starting from the root of the project). Bug: webrtc:8815 Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc Reviewed-on: https://webrtc-review.googlesource.com/46101 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21847}
2018-02-01 14:12:55 +01:00
"codecs/isac/fix/source/structs.h",
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
]
public_configs = [ ":isac_fix_config" ]
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
deps = [
Using fully qualified #include paths in isac code. WebRTC internal code should always used include paths that starts from the root of the project and that clearly identify the header file. This allows 'gn check' to actually keep dependencies under control because 'gn check' cannot enforce anything if the include path is not fully qualified (starting from the root of the project). Bug: webrtc:8815 Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc Reviewed-on: https://webrtc-review.googlesource.com/46101 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21847}
2018-02-01 14:12:55 +01:00
":isac_common",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Using fully qualified #include paths in isac code. WebRTC internal code should always used include paths that starts from the root of the project and that clearly identify the header file. This allows 'gn check' to actually keep dependencies under control because 'gn check' cannot enforce anything if the include path is not fully qualified (starting from the root of the project). Bug: webrtc:8815 Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc Reviewed-on: https://webrtc-review.googlesource.com/46101 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21847}
2018-02-01 14:12:55 +01:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
]
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
}
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
Moving asm code out of isac_fix_c sources list BUG=webrtc:7743 Review-Url: https://codereview.webrtc.org/2973613002 Cr-Commit-Position: refs/heads/master@{#18911}
2017-07-06 03:48:55 -07:00
rtc_source_set("isac_fix_c_arm_asm") {
sources = []
if (current_cpu == "arm" && arm_version >= 7) {
sources += [
"codecs/isac/fix/source/lattice_armv7.S",
"codecs/isac/fix/source/pitch_filter_armv6.S",
]
Adding :isac_fix_c_arm_asm missing dependency. TBR=phoglund@webrtc.org Bug: None Change-Id: I6cb1a442274a627e03a58098d74c8bbf00e492a3 Reviewed-on: https://webrtc-review.googlesource.com/46100 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21806}
2018-01-30 13:36:55 +01:00
deps = [
Adding missing ASM dependencies. Bug: webrtc:8603 Change-Id: I7b417759fcdd01879029afcc5afc50300016fd72 Reviewed-on: https://webrtc-review.googlesource.com/56840 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22159}
2018-02-22 15:10:27 +01:00
":isac_fix_common",
Move asm_defines.h to rtc_base/system/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. Bug: webrtc:8445 NOPRESUBMIT=true Change-Id: I30d01fcb9cbe1427a7703a3cdd7befae751066b5 Reviewed-on: https://webrtc-review.googlesource.com/21982 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22550}
2018-03-22 02:59:49 +01:00
"../../rtc_base/system:asm_defines",
Adding :isac_fix_c_arm_asm missing dependency. TBR=phoglund@webrtc.org Bug: None Change-Id: I6cb1a442274a627e03a58098d74c8bbf00e492a3 Reviewed-on: https://webrtc-review.googlesource.com/46100 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21806}
2018-01-30 13:36:55 +01:00
]
Moving asm code out of isac_fix_c sources list BUG=webrtc:7743 Review-Url: https://codereview.webrtc.org/2973613002 Cr-Commit-Position: refs/heads/master@{#18911}
2017-07-06 03:48:55 -07:00
}
}
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
rtc_source_set("isac_fix_c") {
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
audio_coding: rename interface -> include BUG=webrtc:5095 R=henrik.lundin@webrtc.org TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417173004 . Cr-Commit-Position: refs/heads/master@{#10444}
2015-10-29 11:31:02 +01:00
"codecs/isac/fix/include/audio_decoder_isacfix.h",
"codecs/isac/fix/include/audio_encoder_isacfix.h",
"codecs/isac/fix/include/isacfix.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/arith_routines.c",
"codecs/isac/fix/source/arith_routines_hist.c",
"codecs/isac/fix/source/arith_routines_logist.c",
"codecs/isac/fix/source/arith_routins.h",
"codecs/isac/fix/source/bandwidth_estimator.c",
"codecs/isac/fix/source/bandwidth_estimator.h",
"codecs/isac/fix/source/decode.c",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/isac/fix/source/decode_bwe.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/decode_plc.c",
"codecs/isac/fix/source/encode.c",
"codecs/isac/fix/source/entropy_coding.c",
"codecs/isac/fix/source/filterbank_tables.c",
"codecs/isac/fix/source/filterbank_tables.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/isac/fix/source/filterbanks.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/filters.c",
"codecs/isac/fix/source/initialize.c",
Move AudioDecoderIsac* to its own files Currently, it's sitting in AudioEncoderIsac*'s files, which is less than obvious. This CL puts the encoder and decoder in separate files together with the C implementation; CLs are afoot to make it so for the other built-in codecs as well. BUG=webrtc:4557 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1339253003 . Cr-Commit-Position: refs/heads/master@{#10018}
2015-09-22 19:31:40 +02:00
"codecs/isac/fix/source/isac_fix_type.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/isacfix.c",
"codecs/isac/fix/source/lattice.c",
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
"codecs/isac/fix/source/lattice_c.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/lpc_masking_model.c",
"codecs/isac/fix/source/lpc_masking_model.h",
"codecs/isac/fix/source/lpc_tables.c",
"codecs/isac/fix/source/lpc_tables.h",
"codecs/isac/fix/source/pitch_estimator.c",
"codecs/isac/fix/source/pitch_estimator.h",
Reduce the number of registers used in MIPS optimizations. This change is needed by ChromeOS as it introduces -fno-omit-frame-pointer flag (see code.google.com/p/chromium/issues/detail?id=477749). This causes compile error for MIPS, as some MIPS optimization blocks use maximum possible number of available registers. Also, this change contains minor GN build fix for MIPS platform regarding the pitch_filter_mips.c / pitch_filter_c.c file inclusion. BUG=477749 R=andrew@webrtc.org, djordje.pesut@imgtec.com, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48139004 Patch from Ljubomir Papuga <lpapuga@mips.com>. Cr-Commit-Position: refs/heads/master@{#9047}
2015-04-21 16:52:45 -07:00
"codecs/isac/fix/source/pitch_estimator_c.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/pitch_filter.c",
Reduce the number of registers used in MIPS optimizations. This change is needed by ChromeOS as it introduces -fno-omit-frame-pointer flag (see code.google.com/p/chromium/issues/detail?id=477749). This causes compile error for MIPS, as some MIPS optimization blocks use maximum possible number of available registers. Also, this change contains minor GN build fix for MIPS platform regarding the pitch_filter_mips.c / pitch_filter_c.c file inclusion. BUG=477749 R=andrew@webrtc.org, djordje.pesut@imgtec.com, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48139004 Patch from Ljubomir Papuga <lpapuga@mips.com>. Cr-Commit-Position: refs/heads/master@{#9047}
2015-04-21 16:52:45 -07:00
"codecs/isac/fix/source/pitch_filter_c.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/pitch_gain_tables.c",
"codecs/isac/fix/source/pitch_gain_tables.h",
"codecs/isac/fix/source/pitch_lag_tables.c",
"codecs/isac/fix/source/pitch_lag_tables.h",
"codecs/isac/fix/source/spectrum_ar_model_tables.c",
"codecs/isac/fix/source/spectrum_ar_model_tables.h",
"codecs/isac/fix/source/transform.c",
"codecs/isac/fix/source/transform_tables.c",
]
GN Templates: Move common_inherited_config to the template. Remove common_inherited_config from the targets and add it to the template instead. BUG=webrtc:6187 NOTRY=True Review-Url: https://codereview.webrtc.org/2311843002 Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 06:10:18 -07:00
public_configs = [ ":isac_fix_config" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
deps = [
":isac_common",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:compile_assert_c",
"../../rtc_base:rtc_base_approved",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:sanitizer",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:cpu_features_api",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
]
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
public_deps = [
":isac_fix_common",
]
Fixing cross-compiling issues on android arm BUG=webrtc:7042 Review-Url: https://codereview.webrtc.org/2647293006 Cr-Commit-Position: refs/heads/master@{#16265}
2017-01-25 05:34:46 -08:00
if (rtc_build_with_neon) {
deps += [ ":isac_neon" ]
}
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
if (current_cpu == "arm" && arm_version >= 7) {
sources -= [
"codecs/isac/fix/source/lattice_c.c",
"codecs/isac/fix/source/pitch_filter_c.c",
]
Moving asm code out of isac_fix_c sources list BUG=webrtc:7743 Review-Url: https://codereview.webrtc.org/2973613002 Cr-Commit-Position: refs/heads/master@{#18911}
2017-07-06 03:48:55 -07:00
deps += [ ":isac_fix_c_arm_asm" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530) Includes GN changes from https://webrtc-codereview.appspot.com/39249004/ Android changes for JNI were required due to https://codereview.chromium.org/843103003 Other relevant changes: * src/buildtools: 5c5e924..93b3d0a * src/third_party/boringssl/src: d306f16..b180ee9 * src/third_party/icu: 4e3266f..2081ee6 * src/third_party/libvpx: 5cdd302..33bbffe * src/third_party/usrsctp/usrsctplib: 190c8cb..13718c7 * src/tools/gyp: 4d7c139..3464008 * src/tools/swarming_client: bdad118..1b7bfec Details: https://chromium.googlesource.com/chromium/src/+/b0c3ed3..2c3ffb2/DEPS Clang version was not updated in this roll. R=dpranke@chromium.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40079004 Cr-Commit-Position: refs/heads/master@{#8466} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8466 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 19:08:31 +00:00
if (current_cpu == "mipsel") {
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources += [
"codecs/isac/fix/source/entropy_coding_mips.c",
"codecs/isac/fix/source/filters_mips.c",
"codecs/isac/fix/source/lattice_mips.c",
"codecs/isac/fix/source/pitch_estimator_mips.c",
"codecs/isac/fix/source/transform_mips.c",
]
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
sources -= [
"codecs/isac/fix/source/lattice_c.c",
"codecs/isac/fix/source/pitch_estimator_c.c",
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
if (mips_dsp_rev > 0) {
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
sources += [ "codecs/isac/fix/source/filterbanks_mips.c" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
if (mips_dsp_rev > 1) {
sources += [
"codecs/isac/fix/source/lpc_masking_model_mips.c",
"codecs/isac/fix/source/pitch_filter_mips.c",
]
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
sources -= [ "codecs/isac/fix/source/pitch_filter_c.c" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
}
}
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
if (rtc_build_with_neon) {
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("isac_neon") {
Replace asm NEON function by intrinsics implementation on ARMv7 Passed building isac_neon and modules_unittests on Android ARMv7. Passed modules_unittests with following filters: --gtest_filter=FiltersTest* --gtest_filter=LpcMaskingModelTest* --gtest_filter=TransformTest* --gtest_filter=FilterBanksTest* WebRtcIsacfix_CalculateResidualEnergyNeon is removed, refer more in Issue 4224. The old review url is at: https://webrtc-codereview.appspot.com/37259004/ BUG=4002 R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48319005 Patch from Zhongwei Yao <zhongwei.yao@arm.com>. Change-Id: I4c16e15930f1b3449d67b67bf023fac28121dff8 Cr-Commit-Position: refs/heads/master@{#9140}
2015-05-06 16:39:17 +08:00
sources = [
"codecs/isac/fix/source/entropy_coding_neon.c",
Re-enable WebRtcIsacfix_AllpassFilter2FixDec16Neon WebRtcIsacfix_AllpassFilter2FixDec16Neon was disabled due to a Clang bug. The bug is fixed in current Clang version, re-enable it in this patch. BUG=4567 R=andrew@webrtc.org, kjellander@webrtc.org TEST=buildbot build Change-Id: I71e309cec6caf376181cf9c299c9e8967c9a328e Review URL: https://codereview.webrtc.org/1194773002 . Patch from Zhongwei Yao <zhongwei.yao@arm.com>. Cr-Commit-Position: refs/heads/master@{#9645}
2015-07-28 11:17:40 +08:00
"codecs/isac/fix/source/filterbanks_neon.c",
Replace asm NEON function by intrinsics implementation on ARMv7 Passed building isac_neon and modules_unittests on Android ARMv7. Passed modules_unittests with following filters: --gtest_filter=FiltersTest* --gtest_filter=LpcMaskingModelTest* --gtest_filter=TransformTest* --gtest_filter=FilterBanksTest* WebRtcIsacfix_CalculateResidualEnergyNeon is removed, refer more in Issue 4224. The old review url is at: https://webrtc-codereview.appspot.com/37259004/ BUG=4002 R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48319005 Patch from Zhongwei Yao <zhongwei.yao@arm.com>. Change-Id: I4c16e15930f1b3449d67b67bf023fac28121dff8 Cr-Commit-Position: refs/heads/master@{#9140}
2015-05-06 16:39:17 +08:00
"codecs/isac/fix/source/filters_neon.c",
"codecs/isac/fix/source/lattice_neon.c",
"codecs/isac/fix/source/transform_neon.c",
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
if (current_cpu != "arm64") {
# Enable compilation for the NEON instruction set. This is needed
# since //build/config/arm.gni only enables NEON for iOS, not Android.
# This provides the same functionality as webrtc/build/arm_neon.gypi.
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
cflags = [ "-mfpu=neon" ]
}
# Disable LTO on NEON targets due to compiler bug.
# TODO(fdegans): Enable this. See crbug.com/408997.
GN: Prefix WebRTC specific variables with "rtc_" BUG=3441 TESTED=Trybots + Running GN in a Chromium checkout with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using: --args="os=\"android\" cpu_arch=\"arm\"" R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/27379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:36:10 +00:00
if (rtc_use_lto) {
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
cflags -= [
"-flto",
"-ffat-lto-objects",
]
}
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
":isac_fix_common",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
}
config("pcm16b_config") {
Removing useless include_dirs entry. After the migration from serc/webrtc to src/ this entry in the include_dirs list is not needed anymore. Bug: chromium:611808 Change-Id: I17c87509b73b8a44f758d59ada28d366da664649 Reviewed-on: https://webrtc-review.googlesource.com/1920 Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19894}
2017-09-18 13:09:16 +02:00
include_dirs = [ "codecs/pcm16b/include" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("pcm16b") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
Move AudioDecoderPcm16B next to AudioEncoderPcm16B All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1348113002 . Cr-Commit-Position: refs/heads/master@{#9963}
2015-09-17 07:47:34 +02:00
"codecs/pcm16b/audio_decoder_pcm16b.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/pcm16b/audio_decoder_pcm16b.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/pcm16b/audio_encoder_pcm16b.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/pcm16b/audio_encoder_pcm16b.h",
L16 implementation of the Audio{En,De}coderFactoryTemplate APIs (This is a re-land of https://codereview.webrtc.org/2995523002, with a narrowing conversion removed.) BUG=webrtc:7836, webrtc:7842 Review-Url: https://codereview.webrtc.org/2995993002 Cr-Commit-Position: refs/heads/master@{#19369}
2017-08-16 06:12:57 -07:00
"codecs/pcm16b/pcm16b_common.cc",
"codecs/pcm16b/pcm16b_common.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
Make an AudioEncoder subclass for PCM16B The implementation depends on AudioEncoderPcm to reduce code duplication. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7872 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:47:19 +00:00
deps = [
":g711",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
":legacy_encoded_audio_frame",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Make an AudioEncoder subclass for PCM16B The implementation depends on AudioEncoderPcm to reduce code duplication. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7872 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:47:19 +00:00
]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
public_deps = [
":pcm16b_c",
]
public_configs = [ ":pcm16b_config" ]
}
rtc_source_set("pcm16b_c") {
sources = [
"codecs/pcm16b/pcm16b.c",
"codecs/pcm16b/pcm16b.h",
]
Make an AudioEncoder subclass for PCM16B The implementation depends on AudioEncoderPcm to reduce code duplication. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7872 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:47:19 +00:00
GN Templates: Move common_inherited_config to the template. Remove common_inherited_config from the targets and add it to the template instead. BUG=webrtc:6187 NOTRY=True Review-Url: https://codereview.webrtc.org/2311843002 Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 06:10:18 -07:00
public_configs = [ ":pcm16b_config" ]
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
deps = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("webrtc_opus") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
Move AudioDecoderOpus next to AudioEncoderOpus All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1342933005 . Cr-Commit-Position: refs/heads/master@{#9944}
2015-09-15 17:28:18 +02:00
"codecs/opus/audio_decoder_opus.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/opus/audio_decoder_opus.h",
Make an AudioEncoder subclass for Opus BUG=3926 R=henrik.lundin@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23239004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7552 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 07:28:36 +00:00
"codecs/opus/audio_encoder_opus.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/opus/audio_encoder_opus.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Adding audio network adaptor to AudioEncoderOpus. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2362703002 Cr-Commit-Position: refs/heads/master@{#14555}
2016-10-06 07:13:54 -07:00
":audio_network_adaptor",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Opus implementation of the AudioEncoderFactoryTemplate API This was previously reverted, because external projects were using the internal webrtc::AudioEncoderOpus class and broke when it was renamed. This re-land avoids renaming it immediately, to give those projects time to adapt. It also has to revert some of the changes I had made to the Config struct, since that was also used by the same external projects. BUG=webrtc:7831 Review-Url: https://codereview.webrtc.org/2948483002 Cr-Commit-Position: refs/heads/master@{#18852}
2017-06-30 04:23:22 -07:00
"../../api/audio_codecs/opus:audio_encoder_opus_config",
Update smoothed bitrate. BUG=webrtc:6443 Review-Url: https://codereview.webrtc.org/2546493002 Cr-Commit-Position: refs/heads/master@{#16036}
2017-01-12 10:17:38 -08:00
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../../rtc_base:safe_minmax",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:field_trial_api",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
public_deps = [
":webrtc_opus_c",
Reimplement the builtin audio codec factories using the new stuff in api/ The whole point of all the audio codec stuff we've recently published in api/ is to function as lego bricks so that building stuff like our builtin audio codec factories will be easy. (This has landed once before, but got reverted because of Chromium test failures---apparently, someone isn't ignoring the case of the codec names like they're supposed to. The quick fix was to preserve the same case used by the old implementation.) BUG=webrtc:7821, webrtc:7822 Review-Url: https://codereview.webrtc.org/2998263002 Cr-Commit-Position: refs/heads/master@{#19512}
2017-08-25 03:10:50 -07:00
"../../rtc_base:protobuf_utils",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
]
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
Add 120ms frame ability to ANA BUG=webrtc:7093 Review-Url: https://codereview.webrtc.org/2669733002 Cr-Commit-Position: refs/heads/master@{#16416}
2017-02-02 07:47:19 -08:00
defines = audio_codec_defines
Let Opus increase complexity for low bitrates This change adds code that lets Opus increase the complexity setting at low bitrates (only relevant for mobile where the default complexity is not already maximum). The feature is default off. Also adding a performance test to make sure the complexity adaptation has desired effect. BUG=webrtc:6708 Review-Url: https://codereview.webrtc.org/2503443002 Cr-Commit-Position: refs/heads/master@{#15182}
2016-11-22 02:07:54 -08:00
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
if (rtc_build_opus) {
public_deps += [ rtc_opus_dir ]
} else if (build_with_mozilla) {
Update gn files to support Mozilla build Bug: webrtc:8670 No-Presubmit: true Change-Id: I085dc63daa8274b5068540cbf56b6330f40643fa Reviewed-on: https://webrtc-review.googlesource.com/38920 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21624}
2018-01-15 10:20:00 -05:00
include_dirs = [ "/media/libopus/include" ]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
}
}
rtc_source_set("webrtc_opus_c") {
sources = [
"codecs/opus/opus_inst.h",
"codecs/opus/opus_interface.c",
"codecs/opus/opus_interface.h",
]
Adding build switch for Opus that supports 120ms ptime. BUG=webrtc:7097 TEST=Set "ptime=120", try WebRTC calls over custom build Chromium with and without Opus 120ms. Try both Chromium w <-> Chromium w and Chromium w <-> Chromium w/o Review-Url: https://codereview.webrtc.org/2668633004 Cr-Commit-Position: refs/heads/master@{#16408}
2017-02-01 17:31:11 -08:00
defines = audio_coding_defines
Adding build_opus as a switch in GYP. This is to allow not building Opus. On non-chromium non-gyp chases, one can let WebRTC depend on other Opus builds. BUG= R=kjellander@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43739004 Cr-Commit-Position: refs/heads/master@{#8754} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8754 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 14:04:56 +00:00
if (rtc_build_opus) {
GN: Do not use forward_dependent_configs_from variable. It is deprecated and public_deps should be used instead, which will have the same effect. BUG=None R=brettw@chromium.org,kjellander@webrtc.org Review URL: https://codereview.webrtc.org/1365063004 Cr-Commit-Position: refs/heads/master@{#10074}
2015-09-25 05:57:37 -07:00
public_deps = [
rtc_opus_dir,
]
Adding build_opus as a switch in GYP. This is to allow not building Opus. On non-chromium non-gyp chases, one can let WebRTC depend on other Opus builds. BUG= R=kjellander@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43739004 Cr-Commit-Position: refs/heads/master@{#8754} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8754 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 14:04:56 +00:00
} else if (build_with_mozilla) {
include_dirs = [ getenv("DIST") + "/include/opus" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
deps = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
if (rtc_enable_protobuf) {
proto_library("ana_debug_dump_proto") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
sources = [
"audio_network_adaptor/debug_dump.proto",
]
Add gn dependency between ana_debug_dump_proto and ana_config_proto. BUG=chromium:746106 Review-Url: https://codereview.webrtc.org/2985853002 Cr-Commit-Position: refs/heads/master@{#19158}
2017-07-26 14:49:20 -07:00
deps = [
":ana_config_proto",
]
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
proto_out_dir = "modules/audio_coding/audio_network_adaptor"
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
}
Creating controller manager from config string in audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2364403004 Cr-Commit-Position: refs/heads/master@{#14466}
2016-10-02 14:53:37 -07:00
proto_library("ana_config_proto") {
sources = [
"audio_network_adaptor/config.proto",
]
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
proto_out_dir = "modules/audio_coding/audio_network_adaptor"
Creating controller manager from config string in audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2364403004 Cr-Commit-Position: refs/heads/master@{#14466}
2016-10-02 14:53:37 -07:00
}
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
}
Break the ANA build-target into ANA and ANA-config This is done to solve a dependency-cycle with the RtcEventLog - now the RtcEventLog can depend on the config part of ANA, and be able to peer inside, while the implementation part of ANA can invoke the RtcEventLog. BUG=webrtc:8111 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/3010343002 Cr-Commit-Position: refs/heads/master@{#19793}
2017-09-12 04:38:25 -07:00
rtc_static_library("audio_network_adaptor_config") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Break the ANA build-target into ANA and ANA-config This is done to solve a dependency-cycle with the RtcEventLog - now the RtcEventLog can depend on the config part of ANA, and be able to peer inside, while the implementation part of ANA can invoke the RtcEventLog. BUG=webrtc:8111 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/3010343002 Cr-Commit-Position: refs/heads/master@{#19793}
2017-09-12 04:38:25 -07:00
sources = [
"audio_network_adaptor/audio_network_adaptor_config.cc",
"audio_network_adaptor/include/audio_network_adaptor_config.h",
]
deps = [
"../../api:optional",
]
}
Cleaning build file for audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2399883002 Cr-Commit-Position: refs/heads/master@{#14575}
2016-10-07 07:59:28 -07:00
rtc_static_library("audio_network_adaptor") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Adding AudioNetworkAdaptor interfaces. AudioNetworkAdaptor is supposed to facilitate AudioEncoder to adapt to varying network conditions. This is the first of a sequence of CLs that are to add one implementation of AudioNetworkAdaptor. This CL illustrates the interfaces of the AudioNetworkAdaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2308573002 Cr-Commit-Position: refs/heads/master@{#14115}
2016-09-07 13:51:51 -07:00
sources = [
Adding basic implementation of AudioNetworkAdaptor. The basic implementation of AudioNetworkAdaptor include the introduction of 1. Controller 2. ControllerManager ControllerManager is to hold all needed controllers. It also orders them according to their significance in dealing with current network condition. Controller provides an interface MakeDecision, which has to be implemented by specific controllers. AudioNetworkAdaptorImpl calls MakeDecision of the controllers in the order decided by ControllerManager to collect EncoderRuntimeConfig. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2306083002 Cr-Commit-Position: refs/heads/master@{#14201}
2016-09-13 13:34:15 -07:00
"audio_network_adaptor/audio_network_adaptor_impl.cc",
"audio_network_adaptor/audio_network_adaptor_impl.h",
Relanding of "Adding BitrateController to audio network adaptor." Adding BitrateController to audio network adaptor was first landed in https://codereview.webrtc.org/2334613002/ but reverted in https://codereview.webrtc.org/2352223002/ due to ODR violation. This CL tries to use namespace trick to solve the ODR problem. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2353293002 Cr-Commit-Position: refs/heads/master@{#14343}
2016-09-21 23:01:26 -07:00
"audio_network_adaptor/bitrate_controller.cc",
"audio_network_adaptor/bitrate_controller.h",
Adding ChannelController to audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2319873002 Cr-Commit-Position: refs/heads/master@{#14216}
2016-09-14 06:47:36 -07:00
"audio_network_adaptor/channel_controller.cc",
"audio_network_adaptor/channel_controller.h",
Adding basic implementation of AudioNetworkAdaptor. The basic implementation of AudioNetworkAdaptor include the introduction of 1. Controller 2. ControllerManager ControllerManager is to hold all needed controllers. It also orders them according to their significance in dealing with current network condition. Controller provides an interface MakeDecision, which has to be implemented by specific controllers. AudioNetworkAdaptorImpl calls MakeDecision of the controllers in the order decided by ControllerManager to collect EncoderRuntimeConfig. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2306083002 Cr-Commit-Position: refs/heads/master@{#14201}
2016-09-13 13:34:15 -07:00
"audio_network_adaptor/controller.cc",
"audio_network_adaptor/controller.h",
"audio_network_adaptor/controller_manager.cc",
"audio_network_adaptor/controller_manager.h",
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
"audio_network_adaptor/debug_dump_writer.cc",
"audio_network_adaptor/debug_dump_writer.h",
Adding DTX controller to audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2320093002 Cr-Commit-Position: refs/heads/master@{#14260}
2016-09-16 05:54:39 -07:00
"audio_network_adaptor/dtx_controller.cc",
"audio_network_adaptor/dtx_controller.h",
Reland of "Log audio network adapter decisions in event log." This was originally reviewed https://codereview.webrtc.org/2559953002/ It was reverted due to a bug in the original CL, see https://codereview.webrtc.org/2631703002/ This CL is to fix and reland. BUG=webrtc:6845 Review-Url: https://codereview.webrtc.org/2644863002 Cr-Commit-Position: refs/heads/master@{#16242}
2017-01-24 04:54:59 -08:00
"audio_network_adaptor/event_log_writer.cc",
"audio_network_adaptor/event_log_writer.h",
Introduce FecControllerRplrBased 1. Rename FecController to FecControllerPlrBased. 2. Introduce FecControllerRplrBased - a version of FecController that makes its decision based on RPLR instead of PLR. BUG=webrtc:7058 Review-Url: https://codereview.webrtc.org/2672933003 Cr-Commit-Position: refs/heads/master@{#17373}
2017-03-24 04:12:56 -07:00
"audio_network_adaptor/fec_controller_plr_based.cc",
"audio_network_adaptor/fec_controller_plr_based.h",
"audio_network_adaptor/fec_controller_rplr_based.cc",
"audio_network_adaptor/fec_controller_rplr_based.h",
Adding FrameLengthController to audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2335163002 Cr-Commit-Position: refs/heads/master@{#14339}
2016-09-21 16:00:31 -07:00
"audio_network_adaptor/frame_length_controller.cc",
"audio_network_adaptor/frame_length_controller.h",
Adding AudioNetworkAdaptor interfaces. AudioNetworkAdaptor is supposed to facilitate AudioEncoder to adapt to varying network conditions. This is the first of a sequence of CLs that are to add one implementation of AudioNetworkAdaptor. This CL illustrates the interfaces of the AudioNetworkAdaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2308573002 Cr-Commit-Position: refs/heads/master@{#14115}
2016-09-07 13:51:51 -07:00
"audio_network_adaptor/include/audio_network_adaptor.h",
Avoid code duplication between PLR/RPLR-based FecController BUG=webrtc:7058 Review-Url: https://codereview.webrtc.org/2688613003 Cr-Commit-Position: refs/heads/master@{#17437}
2017-03-29 03:16:58 -07:00
"audio_network_adaptor/util/threshold_curve.h",
Adding AudioNetworkAdaptor interfaces. AudioNetworkAdaptor is supposed to facilitate AudioEncoder to adapt to varying network conditions. This is the first of a sequence of CLs that are to add one implementation of AudioNetworkAdaptor. This CL illustrates the interfaces of the AudioNetworkAdaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2308573002 Cr-Commit-Position: refs/heads/master@{#14115}
2016-09-07 13:51:51 -07:00
]
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
Break the ANA build-target into ANA and ANA-config This is done to solve a dependency-cycle with the RtcEventLog - now the RtcEventLog can depend on the config part of ANA, and be able to peer inside, while the implementation part of ANA can invoke the RtcEventLog. BUG=webrtc:8111 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/3010343002 Cr-Commit-Position: refs/heads/master@{#19793}
2017-09-12 04:38:25 -07:00
public_deps = [
":audio_network_adaptor_config",
]
Adding audio network adaptor to AudioEncoderOpus. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2362703002 Cr-Commit-Position: refs/heads/master@{#14555}
2016-10-06 07:13:54 -07:00
deps = [
"../..:webrtc_common",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. This CL only wires up the new stats but does not set the values yet. This will be added in a follow-up CL. BUG=webrtc:8127 Review-Url: https://codereview.webrtc.org/3011623002 Cr-Commit-Position: refs/heads/master@{#19751}
2017-09-08 08:13:19 -07:00
"../../api/audio_codecs:audio_codecs_api",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../common_audio",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../../logging:rtc_event_audio",
Reland of "Log audio network adapter decisions in event log." This was originally reviewed https://codereview.webrtc.org/2559953002/ It was reverted due to a bug in the original CL, see https://codereview.webrtc.org/2631703002/ This CL is to fix and reland. BUG=webrtc:6845 Review-Url: https://codereview.webrtc.org/2644863002 Cr-Commit-Position: refs/heads/master@{#16242}
2017-01-24 04:54:59 -08:00
"../../logging:rtc_event_log_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
Move file_wrapper.h to rtc_base/system/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 Change-Id: I440974da4d347b09ff042478720d7983056b62b9 Reviewed-on: https://webrtc-review.googlesource.com/21226 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22579}
2018-03-23 10:39:34 +01:00
"../../rtc_base/system:file_wrapper",
Adding audio network adaptor to AudioEncoderOpus. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2362703002 Cr-Commit-Position: refs/heads/master@{#14555}
2016-10-06 07:13:54 -07:00
"../../system_wrappers",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:field_trial_api",
Adding audio network adaptor to AudioEncoderOpus. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2362703002 Cr-Commit-Position: refs/heads/master@{#14555}
2016-10-06 07:13:54 -07:00
]
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
if (rtc_enable_protobuf) {
Adding audio network adaptor to AudioEncoderOpus. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2362703002 Cr-Commit-Position: refs/heads/master@{#14555}
2016-10-06 07:13:54 -07:00
deps += [
Creating controller manager from config string in audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2364403004 Cr-Commit-Position: refs/heads/master@{#14466}
2016-10-02 14:53:37 -07:00
":ana_config_proto",
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
":ana_debug_dump_proto",
]
}
Reland of "Log audio network adapter decisions in event log." This was originally reviewed https://codereview.webrtc.org/2559953002/ It was reverted due to a bug in the original CL, see https://codereview.webrtc.org/2631703002/ This CL is to fix and reland. BUG=webrtc:6845 Review-Url: https://codereview.webrtc.org/2644863002 Cr-Commit-Position: refs/heads/master@{#16242}
2017-01-24 04:54:59 -08:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
Adding AudioNetworkAdaptor interfaces. AudioNetworkAdaptor is supposed to facilitate AudioEncoder to adapt to varying network conditions. This is the first of a sequence of CLs that are to add one implementation of AudioNetworkAdaptor. This CL illustrates the interfaces of the AudioNetworkAdaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2308573002 Cr-Commit-Position: refs/heads/master@{#14115}
2016-09-07 13:51:51 -07:00
}
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
rtc_source_set("neteq_decoder_enum") {
sources = [
"neteq/neteq_decoder_enum.cc",
"neteq/neteq_decoder_enum.h",
]
deps = [
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
"../../api/audio_codecs:audio_codecs_api",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
]
}
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
rtc_static_library("neteq") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
"neteq/accelerate.cc",
"neteq/accelerate.h",
"neteq/audio_multi_vector.cc",
"neteq/audio_multi_vector.h",
"neteq/audio_vector.cc",
"neteq/audio_vector.h",
"neteq/background_noise.cc",
"neteq/background_noise.h",
"neteq/buffer_level_filter.cc",
"neteq/buffer_level_filter.h",
"neteq/comfort_noise.cc",
"neteq/comfort_noise.h",
Reland "Avoiding overflow in cross correlation in NetEq." The original CL is https://codereview.webrtc.org/1908623002/ An error was caused by that and this CL fix that problem and reland the CL. BUG= Review-Url: https://codereview.webrtc.org/1931933004 Cr-Commit-Position: refs/heads/master@{#12589}
2016-05-02 01:50:30 -07:00
"neteq/cross_correlation.cc",
"neteq/cross_correlation.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"neteq/decision_logic.cc",
"neteq/decision_logic.h",
"neteq/decision_logic_fax.cc",
"neteq/decision_logic_fax.h",
"neteq/decision_logic_normal.cc",
"neteq/decision_logic_normal.h",
"neteq/decoder_database.cc",
"neteq/decoder_database.h",
"neteq/defines.h",
"neteq/delay_manager.cc",
"neteq/delay_manager.h",
"neteq/delay_peak_detector.cc",
"neteq/delay_peak_detector.h",
"neteq/dsp_helper.cc",
"neteq/dsp_helper.h",
"neteq/dtmf_buffer.cc",
"neteq/dtmf_buffer.h",
"neteq/dtmf_tone_generator.cc",
"neteq/dtmf_tone_generator.h",
"neteq/expand.cc",
"neteq/expand.h",
audio_coding: rename interface -> include BUG=webrtc:5095 R=henrik.lundin@webrtc.org TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417173004 . Cr-Commit-Position: refs/heads/master@{#10444}
2015-10-29 11:31:02 +01:00
"neteq/include/neteq.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"neteq/merge.cc",
"neteq/merge.h",
NetEq: Rename Nack to NackTracker to avoid name collisions in GN BUG=webrtc:5949 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2045243002 Cr-Commit-Position: refs/heads/master@{#13069}
2016-06-08 06:43:41 -07:00
"neteq/nack_tracker.cc",
"neteq/nack_tracker.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"neteq/neteq.cc",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"neteq/neteq_impl.cc",
"neteq/neteq_impl.h",
"neteq/normal.cc",
"neteq/normal.h",
NetEq: Use TickTimer in PacketBuffer This change makes use of the TickTimer::Stopwatch in Packets. When a packet is inserted into the PacketBuffer, a Stopwatch object is attached to it. When the packet is extracted from the buffer, the Stopwatch is read to know how long the packet waited in the buffer. BUG=webrtc:5608 Review URL: https://codereview.webrtc.org/1917913002 Cr-Commit-Position: refs/heads/master@{#12508}
2016-04-26 07:45:16 -07:00
"neteq/packet.cc",
"neteq/packet.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"neteq/packet_buffer.cc",
"neteq/packet_buffer.h",
"neteq/post_decode_vad.cc",
"neteq/post_decode_vad.h",
"neteq/preemptive_expand.cc",
"neteq/preemptive_expand.h",
"neteq/random_vector.cc",
"neteq/random_vector.h",
Moved Opus-specific payload splitting into AudioDecoderOpus. The biggest change to NetEq is the move from a primary flag, to a Priority with two separate levels: one set by RED splitting and one set by the codec itself. This allows us to unambigously prioritize "fallback" packets from these two sources. I've chosen what I believe is the sensible ordering: packets that the codec prioritizes are chosen first, regardless of if they are secondary RED packets or not. So if we were to use Opus w/ FEC in RED, we'd only do Opus FEC decoding if there was no RED packet that could cover the time slot. With this change, PayloadSplitter now only deals with RED packets. Maybe it should be renamed RedPayloadSplitter? BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2342443005 Cr-Commit-Position: refs/heads/master@{#14347}
2016-09-22 02:06:28 -07:00
"neteq/red_payload_splitter.cc",
"neteq/red_payload_splitter.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"neteq/rtcp.cc",
"neteq/rtcp.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"neteq/statistics_calculator.cc",
"neteq/statistics_calculator.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"neteq/sync_buffer.cc",
"neteq/sync_buffer.h",
Add a new TickTimer class to NetEq The new class is intended to be used as a central time-keeping object inside NetEq. The actual use of the class will come in subsequent changes. BUG=webrtc:5608 Review URL: https://codereview.webrtc.org/1910523003 Cr-Commit-Position: refs/heads/master@{#12477}
2016-04-22 13:21:43 -07:00
"neteq/tick_timer.cc",
"neteq/tick_timer.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"neteq/time_stretch.cc",
"neteq/time_stretch.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"neteq/timestamp_scaler.cc",
"neteq/timestamp_scaler.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
deps = [
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
":audio_coding_module_typedefs",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
":cng",
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
":neteq_decoder_enum",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Fix style violations in common_types.h and config.h Mostly, it's about moving constructors and descructors to the .cc files, so that they won't be inlined everywhere. The reason this CL is so big is that a lot of code was using common_types.h without declaring a dependency on webrtc_common, which broke the build once common_types.h started to depend on common_types.cc. BUG=163 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26089004 Cr-Commit-Position: refs/heads/master@{#8516} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 13:59:22 +00:00
"../..:webrtc_common",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../../api:libjingle_peerconnection_api",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
This CL removes all usages of our custom ostream << overloads. This prepares us for removing them altogether. Bug: webrtc:8982 Change-Id: I66002cc8d4bf0e07925766d568d2498422f0f38e Reviewed-on: https://webrtc-review.googlesource.com/64142 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Henrik Grunell <henrikg@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22707}
2018-04-03 13:40:05 +02:00
"../../rtc_base:audio_format_to_string",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:gtest_prod",
"../../rtc_base:rtc_base_approved",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../../rtc_base:safe_minmax",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:sanitizer",
Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix Bug: chromium:805946 Change-Id: Ibb5dce9af27d0e48c9aee6b0a860b6f62b3c76a0 Reviewed-on: https://webrtc-review.googlesource.com/46145 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21889}
2018-02-05 10:33:35 +01:00
"../../rtc_base/system:fallthrough",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
# Although providing only test support, this target must be outside of the
# rtc_include_tests conditional. The reason is that it supports fuzzer tests
# that ultimately are built and run as a part of the Chromium ecosystem, which
# does not set the rtc_include_tests flag.
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
rtc_source_set("neteq_tools_minimal") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
sources = [
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
"neteq/tools/audio_sink.cc",
"neteq/tools/audio_sink.h",
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
"neteq/tools/encode_neteq_input.cc",
"neteq/tools/encode_neteq_input.h",
Add NetEqInput::PacketData::ToString method This new method prints information about the packet. BUG=webrtc:7467 Review-Url: https://codereview.webrtc.org/2844283002 Cr-Commit-Position: refs/heads/master@{#17922}
2017-04-28 01:35:53 -07:00
"neteq/tools/neteq_input.cc",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
"neteq/tools/neteq_input.h",
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
"neteq/tools/neteq_test.cc",
"neteq/tools/neteq_test.h",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
"neteq/tools/packet.cc",
"neteq/tools/packet.h",
"neteq/tools/packet_source.cc",
"neteq/tools/packet_source.h",
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
deps = [
":neteq",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
"..:module_api",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../../api:libjingle_peerconnection_api",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../../api/audio_codecs:audio_codecs_api",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:builtin_audio_decoder_factory",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
"../rtp_rtcp",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
]
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
}
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
rtc_source_set("neteq_test_tools") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
testonly = true
sources = [
"neteq/tools/audio_checksum.h",
"neteq/tools/audio_loop.cc",
"neteq/tools/audio_loop.h",
"neteq/tools/constant_pcm_packet_source.cc",
"neteq/tools/constant_pcm_packet_source.h",
"neteq/tools/output_audio_file.h",
"neteq/tools/output_wav_file.h",
"neteq/tools/rtp_file_source.cc",
"neteq/tools/rtp_file_source.h",
"neteq/tools/rtp_generator.cc",
"neteq/tools/rtp_generator.h",
]
public_configs = [ ":neteq_tools_config" ]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
":pcm16b",
"..:module_api",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../..:webrtc_common",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../../api:libjingle_peerconnection_api",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Use generic MessageDigest class instead of MD5 / SHA-1 specific classes. This allows removing the specific classes in a later CL. Bug: webrtc:8677 Change-Id: I3b9c1f3191c38e6d31a3de990e2d882505e79adc Reviewed-on: https://webrtc-review.googlesource.com/35040 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Joachim Bauch <jbauch@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21412}
2017-12-19 22:27:51 +01:00
"../../rtc_base:rtc_base",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_base_tests_utils",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../../test:rtp_test_utils",
"../rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
public_deps = [
":neteq_tools",
":neteq_tools_minimal",
]
if (rtc_enable_protobuf) {
sources += [
"neteq/tools/neteq_packet_source_input.cc",
"neteq/tools/neteq_packet_source_input.h",
]
deps += [ ":rtc_event_log_source" ]
}
}
config("neteq_tools_config") {
include_dirs = [ "tools" ]
}
rtc_source_set("neteq_tools") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
sources = [
"neteq/tools/fake_decode_from_file.cc",
"neteq/tools/fake_decode_from_file.h",
Add NetEq delay plotting to event_log_visualizer This CL adds the capability to analyze and plot how NetEq behaves in response to a network trace. BUG=webrtc:7467 Review-Url: https://codereview.webrtc.org/2876423002 Cr-Commit-Position: refs/heads/master@{#18590}
2017-06-14 06:09:58 -07:00
"neteq/tools/neteq_delay_analyzer.cc",
"neteq/tools/neteq_delay_analyzer.h",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"neteq/tools/neteq_replacement_input.cc",
"neteq/tools/neteq_replacement_input.h",
]
public_configs = [ ":neteq_tools_config" ]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../..:webrtc_common",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../../api/audio_codecs:audio_codecs_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
public_deps = [
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
":neteq_input_audio_tools",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
":neteq_tools_minimal",
]
}
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
rtc_source_set("neteq_input_audio_tools") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
sources = [
"neteq/tools/input_audio_file.cc",
"neteq/tools/input_audio_file.h",
"neteq/tools/resample_input_audio_file.cc",
"neteq/tools/resample_input_audio_file.h",
]
deps = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
"../..:webrtc_common",
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
"../../rtc_base:rtc_base_approved",
]
}
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
if (rtc_enable_protobuf) {
rtc_static_library("rtc_event_log_source") {
testonly = true
sources = [
"neteq/tools/rtc_event_log_source.cc",
"neteq/tools/rtc_event_log_source.h",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
Fix circular dependency in rtc_event_log. Bug: webrtc:6828 Change-Id: Ief948b6799455cfda6cb89e2e632f5fd42df0881 Reviewed-on: https://webrtc-review.googlesource.com/25840 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20928}
2017-11-24 13:51:52 +01:00
":neteq_tools_minimal",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../../logging:rtc_event_log_parser",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Stop using public_deps in logging/. Bug: webrtc:8603 Change-Id: Id0df997620a27e47067e4b21e4e8db16aec90640 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30940 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21228}
2017-12-12 12:46:13 +01:00
"../rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
public_deps = [
"../../logging:rtc_event_log_proto",
]
}
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (rtc_include_tests) {
Add missing files to audio_coding Bug: webrtc:7650 Change-Id: I8d7d8c3998799404ec6283896883d195468cdfdc Reviewed-on: https://webrtc-review.googlesource.com/37622 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21502}
2018-01-05 11:46:26 +01:00
rtc_source_set("mocks") {
testonly = true
sources = [
"audio_network_adaptor/mock/mock_audio_network_adaptor.h",
"audio_network_adaptor/mock/mock_controller.h",
"audio_network_adaptor/mock/mock_controller_manager.h",
"audio_network_adaptor/mock/mock_debug_dump_writer.h",
]
deps = [
":audio_network_adaptor",
"../../test:test_support",
]
}
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
group("audio_coding_tests") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
testonly = true
public_deps = [
":acm_receive_test",
":acm_send_test",
":audio_codec_speed_tests",
":audio_decoder_unittests",
":audio_decoder_unittests",
":delay_test",
":g711_test",
":g722_test",
":ilbc_test",
":insert_packet_with_timing",
":isac_api_test",
":isac_fix_test",
":isac_switch_samprate_test",
":isac_test",
":neteq_ilbc_quality_test",
":neteq_isac_quality_test",
":neteq_opus_quality_test",
Add pcm16b quality test supporting 48khz. Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Bug: webrtc:8836 Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Reviewed-on: https://webrtc-review.googlesource.com/47400 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21878}
2018-02-02 17:21:46 +01:00
":neteq_pcm16b_quality_test",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
":neteq_pcmu_quality_test",
":neteq_speed_test",
":rtp_analyze",
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
":rtp_encode",
Replacing the legacy tool RTPjitter with a new rtp_jitter This new tool provides the similar functionality as the legacy tool, but is implemented using less legacy helpers. It also replaces RTPtimeshift and RTPchange. The most significant change versus the old RTPjitter tool is that the new tool takes the timing data in the form of integers in a text file (instead of the binary data file used by the old tool). This should make it easier to create custom timing files when needed. Bug: webrtc:2692 Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287 Reviewed-on: https://webrtc-review.googlesource.com/25700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:39:39 +01:00
":rtp_jitter",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
":rtpcat",
":webrtc_opus_fec_test",
]
if (rtc_enable_protobuf) {
public_deps += [ ":neteq_rtpplay" ]
}
}
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
rtc_source_set("audio_coding_modules_tests") {
testonly = true
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
sources = [
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/ACMTest.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/APITest.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/APITest.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/Channel.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/Channel.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/EncodeDecodeTest.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/EncodeDecodeTest.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/PCMFile.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/PCMFile.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/PacketLossTest.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/PacketLossTest.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/RTPFile.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/RTPFile.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/TestAllCodecs.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/TestAllCodecs.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/TestRedFec.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/TestRedFec.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/TestStereo.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/TestStereo.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/TestVADDTX.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/TestVADDTX.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/Tester.cc",
"test/TwoWayCommunication.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/TwoWayCommunication.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/iSACTest.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/iSACTest.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/opus_test.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/opus_test.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/target_delay_unittest.cc",
"test/utility.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/utility.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
]
deps = [
":audio_coding",
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
":audio_coding_module_typedefs",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
":audio_format_conversion",
":pcm16b_c",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"../..:webrtc_common",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:builtin_audio_decoder_factory",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Move rw_lock_wrapper.h to rtc_base/synchronization/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 NOPRESUBMIT=true Change-Id: Ie2879aca5fc1667e4222499d2a8fc2bba9ae2425 Reviewed-on: https://webrtc-review.googlesource.com/21328 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22587}
2018-03-23 14:53:54 +01:00
"../../rtc_base/synchronization:rw_lock_wrapper",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"../../test:test_support",
]
defines = audio_coding_defines
if (is_win) {
cflags = [
# TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
"/wd4373", # virtual function override.
]
}
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
rtc_source_set("audio_coding_perf_tests") {
testonly = true
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
sources = [
"codecs/opus/opus_complexity_unittest.cc",
"neteq/test/neteq_performance_unittest.cc",
]
deps = [
":neteq_test_support",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_test_tools",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
"../..:webrtc_common",
Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix We've done this previously with the other audio encoders, but Opus had to wait until all external users had been updated. BUG=webrtc:7847 Change-Id: I70422d7b6c715f32a43bee88febcf6b6155e18b3 Reviewed-on: https://webrtc-review.googlesource.com/8000 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20424}
2017-10-25 09:57:40 +02:00
"../../api/audio_codecs/opus:audio_encoder_opus",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
"../../system_wrappers:field_trial_api",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Reland "iOS: Save perf results under Documents/perf_result.json" This will require a manual roll to downstream projects, since the //test:perf_test target was introduced. This is a reland of 10a8e7a9b5261a7e3ce19900ba3511be3b5911f8 Original change's description: > iOS: Save perf results under Documents/perf_result.json > > TBR=henrika@webrtc.org > > Bug: webrtc:7156 > Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5 > Reviewed-on: https://webrtc-review.googlesource.com/29202 > Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21244} R=henrika@webrtc.org, phoglund@webrtc.org Bug: webrtc:7156 Change-Id: I85fc7bc5fce0894af90017b71b9952b61b523424 Reviewed-on: https://webrtc-review.googlesource.com/37643 Reviewed-by: Patrik Höglund <phoglund@google.com> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21518}
2018-01-05 15:34:09 +01:00
"../../test:perf_test",
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
"../../test:test_support",
]
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ ) Reason for revert: Trying to re-land after solving some related issues. There are no changes compared to the original CL. Original issue's description: > Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ ) > > Reason for revert: > I will try to reland next week because it is causing some problems. > > Original issue's description: > > To accommodate some downstream WebRTC users we need to loosen > > the coupling between our code and the //third_party/protobuf. > > > > This includes using typedefs to define strings instead of > > assuming std::string. > > > > After this refactoring it will be possible to link with other > > protobuf implementations than the current one. > > > > We moved the PRESUBMIT check to another CL [1]. The goal of this > > presubmit is to avoid the direct usage of google::protobuf outside > > of the webrtc/base/protobuf_utils.h header file. > > > > [1] - https://codereview.webrtc.org/2753823003/ > > > > BUG=webrtc:7340 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2747863003 > > Cr-Commit-Position: refs/heads/master@{#17466} > > Committed: https://chromium.googlesource.com/external/webrtc/+/16ab93b952f9e8268f2e663ffe49548e8043d5af > > TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:7340 > > Review-Url: https://codereview.webrtc.org/2786363002 > Cr-Commit-Position: refs/heads/master@{#17483} > Committed: https://chromium.googlesource.com/external/webrtc/+/d00aad5eb2fa5a7b5aeda714f7702b50cd26ee28 TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7340 NOTRY=True Review-Url: https://codereview.webrtc.org/2791963003 Cr-Commit-Position: refs/heads/master@{#17584}
2017-04-07 00:59:12 -07:00
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_source_set("acm_receive_test") {
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
testonly = true
sources = [
Drop _oldapi from ACM test file names Updating GN files, include paths, and include guards BUG=None NOTRY=True NOPRESUBMIT=true Review-Url: https://codereview.webrtc.org/2387113005 Cr-Commit-Position: refs/heads/master@{#14542}
2016-10-06 01:31:32 -07:00
"acm2/acm_receive_test.cc",
"acm2/acm_receive_test.h",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
]
defines = audio_coding_defines
deps = audio_coding_deps + [
":audio_coding",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
":audio_format_conversion",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_tools",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../test:test_support",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"//testing/gtest",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_source_set("acm_send_test") {
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
testonly = true
sources = [
Drop _oldapi from ACM test file names Updating GN files, include paths, and include guards BUG=None NOTRY=True NOPRESUBMIT=true Review-Url: https://codereview.webrtc.org/2387113005 Cr-Commit-Position: refs/heads/master@{#14542}
2016-10-06 01:31:32 -07:00
"acm2/acm_send_test.cc",
"acm2/acm_send_test.h",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
]
defines = audio_coding_defines
deps = audio_coding_deps + [
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
":audio_coding",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_tools",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../../api/audio_codecs:audio_codecs_api",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../test:test_support",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"//testing/gtest",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("delay_test") {
Target delay_test migrated to GN. Review-Url: https://codereview.webrtc.org/2188473003 Cr-Commit-Position: refs/heads/master@{#13547}
2016-07-28 01:21:29 -07:00
testonly = true
sources = [
"test/Channel.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/Channel.h",
Target delay_test migrated to GN. Review-Url: https://codereview.webrtc.org/2188473003 Cr-Commit-Position: refs/heads/master@{#13547}
2016-07-28 01:21:29 -07:00
"test/PCMFile.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/PCMFile.h",
Target delay_test migrated to GN. Review-Url: https://codereview.webrtc.org/2188473003 Cr-Commit-Position: refs/heads/master@{#13547}
2016-07-28 01:21:29 -07:00
"test/delay_test.cc",
"test/utility.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/utility.h",
Target delay_test migrated to GN. Review-Url: https://codereview.webrtc.org/2188473003 Cr-Commit-Position: refs/heads/master@{#13547}
2016-07-28 01:21:29 -07:00
]
deps = [
":audio_coding",
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
":audio_coding_module_typedefs",
Stop using old AudioCodingModule::RegisterReceiveCodec overloads BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/2388153004 Cr-Commit-Position: refs/heads/master@{#14753}
2016-10-24 13:47:09 -07:00
":audio_format_conversion",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Target delay_test migrated to GN. Review-Url: https://codereview.webrtc.org/2188473003 Cr-Commit-Position: refs/heads/master@{#13547}
2016-07-28 01:21:29 -07:00
"../../:webrtc_common",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Target delay_test migrated to GN. Review-Url: https://codereview.webrtc.org/2188473003 Cr-Commit-Position: refs/heads/master@{#13547}
2016-07-28 01:21:29 -07:00
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Target delay_test migrated to GN. Review-Url: https://codereview.webrtc.org/2188473003 Cr-Commit-Position: refs/heads/master@{#13547}
2016-07-28 01:21:29 -07:00
"../../test:test_support",
"../rtp_rtcp",
"//testing/gtest",
]
} # delay_test
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("insert_packet_with_timing") {
Migrated GN target insert_packet_with_timing. Review-Url: https://codereview.webrtc.org/2187003002 Cr-Commit-Position: refs/heads/master@{#13548}
2016-07-28 01:27:10 -07:00
testonly = true
sources = [
"test/Channel.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/Channel.h",
Migrated GN target insert_packet_with_timing. Review-Url: https://codereview.webrtc.org/2187003002 Cr-Commit-Position: refs/heads/master@{#13548}
2016-07-28 01:27:10 -07:00
"test/PCMFile.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/PCMFile.h",
Migrated GN target insert_packet_with_timing. Review-Url: https://codereview.webrtc.org/2187003002 Cr-Commit-Position: refs/heads/master@{#13548}
2016-07-28 01:27:10 -07:00
"test/insert_packet_with_timing.cc",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
Migrated GN target insert_packet_with_timing. Review-Url: https://codereview.webrtc.org/2187003002 Cr-Commit-Position: refs/heads/master@{#13548}
2016-07-28 01:27:10 -07:00
}
deps = [
":audio_coding",
Stop using old AudioCodingModule::RegisterReceiveCodec overloads BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/2388153004 Cr-Commit-Position: refs/heads/master@{#14753}
2016-10-24 13:47:09 -07:00
":audio_format_conversion",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Migrated GN target insert_packet_with_timing. Review-Url: https://codereview.webrtc.org/2187003002 Cr-Commit-Position: refs/heads/master@{#13548}
2016-07-28 01:27:10 -07:00
"../../:webrtc_common",
Move optional.h to webrtc/api/ We use Optional in our public API, so its header should be in webrtc/api/. BUG=webrtc:8205 Review-Url: https://codereview.webrtc.org/3011943002 Cr-Commit-Position: refs/heads/master@{#19693}
2017-09-05 08:43:13 -07:00
"../../api:optional",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Migrated GN target insert_packet_with_timing. Review-Url: https://codereview.webrtc.org/2187003002 Cr-Commit-Position: refs/heads/master@{#13548}
2016-07-28 01:27:10 -07:00
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Migrated GN target insert_packet_with_timing. Review-Url: https://codereview.webrtc.org/2187003002 Cr-Commit-Position: refs/heads/master@{#13548}
2016-07-28 01:27:10 -07:00
"../../test:test_support",
"../rtp_rtcp",
"//testing/gtest",
]
} # insert_packet_with_timing
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
audio_decoder_unittests_resources =
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
[ "../../resources/audio_coding/testfile32kHz.pcm" ]
GN: Fix resource files for iOS test target In order to get resource files to be properly packaged into the .app for a unit test on iOS, the resource files needs to be listed as sources in a bundle_data target. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2292853002 Cr-Commit-Position: refs/heads/master@{#13968}
2016-08-30 02:53:49 -07:00
if (is_ios) {
bundle_data("audio_decoder_unittests_bundle_data") {
testonly = true
sources = audio_decoder_unittests_resources
outputs = [
"{{bundle_resources_dir}}/{{source_file_part}}",
]
}
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_test("audio_decoder_unittests") {
GN conversion of audio_decoder_unittests BUG=webrtc 6036 Review-Url: https://codereview.webrtc.org/2180853006 Cr-Commit-Position: refs/heads/master@{#13588}
2016-08-01 07:49:42 -07:00
testonly = true
sources = [
"neteq/audio_decoder_unittest.cc",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN conversion of audio_decoder_unittests BUG=webrtc 6036 Review-Url: https://codereview.webrtc.org/2180853006 Cr-Commit-Position: refs/heads/master@{#13588}
2016-08-01 07:49:42 -07:00
}
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
deps = [
"../../test:fileutils",
]
GN conversion of audio_decoder_unittests BUG=webrtc 6036 Review-Url: https://codereview.webrtc.org/2180853006 Cr-Commit-Position: refs/heads/master@{#13588}
2016-08-01 07:49:42 -07:00
defines = neteq_defines
deps += audio_coding_deps
deps += [
Add missing dependency to audio_decoder_unittests. BUG=None NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2684573002 Cr-Commit-Position: refs/heads/master@{#16470}
2017-02-07 06:26:29 -08:00
":ilbc",
GN conversion of audio_decoder_unittests BUG=webrtc 6036 Review-Url: https://codereview.webrtc.org/2180853006 Cr-Commit-Position: refs/heads/master@{#13588}
2016-08-01 07:49:42 -07:00
":isac",
":isac_fix",
":neteq",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_tools",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Opus implementation of the AudioEncoderFactoryTemplate API This was previously reverted, because external projects were using the internal webrtc::AudioEncoderOpus class and broke when it was renamed. This re-land avoids renaming it immediately, to give those projects time to adapt. It also has to revert some of the changes I had made to the Config struct, since that was also used by the same external projects. BUG=webrtc:7831 Review-Url: https://codereview.webrtc.org/2948483002 Cr-Commit-Position: refs/heads/master@{#18852}
2017-06-30 04:23:22 -07:00
"../../api/audio_codecs/opus:audio_encoder_opus",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
"../../common_audio",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
GN conversion of audio_decoder_unittests BUG=webrtc 6036 Review-Url: https://codereview.webrtc.org/2180853006 Cr-Commit-Position: refs/heads/master@{#13588}
2016-08-01 07:49:42 -07:00
"//testing/gtest",
]
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
data = audio_decoder_unittests_resources
GN conversion of audio_decoder_unittests BUG=webrtc 6036 Review-Url: https://codereview.webrtc.org/2180853006 Cr-Commit-Position: refs/heads/master@{#13588}
2016-08-01 07:49:42 -07:00
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
GN: Update tests to have the correct shard timeout value on Android. TBR=mflodman@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2231413002 Cr-Commit-Position: refs/heads/master@{#13741}
2016-08-15 02:29:11 -07:00
shard_timeout = 900
GN conversion of audio_decoder_unittests BUG=webrtc 6036 Review-Url: https://codereview.webrtc.org/2180853006 Cr-Commit-Position: refs/heads/master@{#13588}
2016-08-01 07:49:42 -07:00
}
GN: Fix resource files for iOS test target In order to get resource files to be properly packaged into the .app for a unit test on iOS, the resource files needs to be listed as sources in a bundle_data target. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2292853002 Cr-Commit-Position: refs/heads/master@{#13968}
2016-08-30 02:53:49 -07:00
if (is_ios) {
deps += [ ":audio_decoder_unittests_bundle_data" ]
GN conversion of audio_decoder_unittests BUG=webrtc 6036 Review-Url: https://codereview.webrtc.org/2180853006 Cr-Commit-Position: refs/heads/master@{#13588}
2016-08-01 07:49:42 -07:00
}
} # audio_decoder_unittests
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (rtc_enable_protobuf) {
proto_library("neteq_unittest_proto") {
sources = [
"neteq/neteq_unittest.proto",
]
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
proto_out_dir = "modules/audio_coding/neteq"
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
GN: Add neteq_rtpplay and rtc_event_log_source BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2081113003 Cr-Commit-Position: refs/heads/master@{#13239}
2016-06-21 05:38:42 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_test("neteq_rtpplay") {
GN: Add neteq_rtpplay and rtc_event_log_source BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2081113003 Cr-Commit-Position: refs/heads/master@{#13239}
2016-06-21 05:38:42 -07:00
testonly = true
defines = []
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
deps = [
"..:module_api",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
]
GN: Add neteq_rtpplay and rtc_event_log_source BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2081113003 Cr-Commit-Position: refs/heads/master@{#13239}
2016-06-21 05:38:42 -07:00
sources = [
"neteq/tools/neteq_rtpplay.cc",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Add neteq_rtpplay and rtc_event_log_source BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2081113003 Cr-Commit-Position: refs/heads/master@{#13239}
2016-06-21 05:38:42 -07:00
}
if (is_win) {
cflags = [
# TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
"/wd4373", # virtual function override.
]
}
deps += [
":neteq",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_test_tools",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
GN: Add neteq_rtpplay and rtc_event_log_source BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2081113003 Cr-Commit-Position: refs/heads/master@{#13239}
2016-06-21 05:38:42 -07:00
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
]
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
Adding missing resources to audio_codec_speed_tests. BUG=none Review-Url: https://codereview.webrtc.org/2727973004 Cr-Commit-Position: refs/heads/master@{#19168}
2017-07-27 05:49:57 -07:00
audio_codec_speed_tests_resources = [
"//resources/audio_coding/music_stereo_48kHz.pcm",
"//resources/audio_coding/speech_mono_16kHz.pcm",
"//resources/audio_coding/speech_mono_32_48kHz.pcm",
]
if (is_ios) {
bundle_data("audio_codec_speed_tests_data") {
testonly = true
sources = audio_codec_speed_tests_resources
outputs = [
"{{bundle_resources_dir}}/{{source_file_part}}",
]
}
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_test("audio_codec_speed_tests") {
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
testonly = true
defines = []
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
deps = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
]
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
sources = [
"codecs/isac/fix/test/isac_speed_test.cc",
"codecs/opus/opus_speed_test.cc",
"codecs/tools/audio_codec_speed_test.cc",
"codecs/tools/audio_codec_speed_test.h",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
}
Adding missing resources to audio_codec_speed_tests. BUG=none Review-Url: https://codereview.webrtc.org/2727973004 Cr-Commit-Position: refs/heads/master@{#19168}
2017-07-27 05:49:57 -07:00
data = audio_codec_speed_tests_resources
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
GN: Update tests to have the correct shard timeout value on Android. TBR=mflodman@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2231413002 Cr-Commit-Position: refs/heads/master@{#13741}
2016-08-15 02:29:11 -07:00
shard_timeout = 900
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
}
Adding missing resources to audio_codec_speed_tests. BUG=none Review-Url: https://codereview.webrtc.org/2727973004 Cr-Commit-Position: refs/heads/master@{#19168}
2017-07-27 05:49:57 -07:00
if (is_ios) {
deps += [ ":audio_codec_speed_tests_data" ]
}
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
deps += [
":isac_fix",
":webrtc_opus",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Move RtpExtension to api/ directory and config.h/.cc to call/. BUG=webrtc:5876 R=deadbeef@webrtc.org, solenberg@webrtc.org Review-Url: https://codereview.webrtc.org/3004723002 . Cr-Commit-Position: refs/heads/master@{#19639}
2017-09-01 15:29:28 +02:00
"../../api:libjingle_peerconnection_api",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Move RtpExtension to api/ directory and config.h/.cc to call/. BUG=webrtc:5876 R=deadbeef@webrtc.org, solenberg@webrtc.org Review-Url: https://codereview.webrtc.org/3004723002 . Cr-Commit-Position: refs/heads/master@{#19639}
2017-09-01 15:29:28 +02:00
"../../system_wrappers:metrics_default",
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
"../../system_wrappers:system_wrappers_default",
Move RtpExtension to api/ directory and config.h/.cc to call/. BUG=webrtc:5876 R=deadbeef@webrtc.org, solenberg@webrtc.org Review-Url: https://codereview.webrtc.org/3004723002 . Cr-Commit-Position: refs/heads/master@{#19639}
2017-09-01 15:29:28 +02:00
"../../test:field_trial",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
"../audio_processing",
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
"//testing/gtest",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_source_set("neteq_test_support") {
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
testonly = true
sources = [
"neteq/tools/neteq_external_decoder_test.cc",
"neteq/tools/neteq_external_decoder_test.h",
"neteq/tools/neteq_performance_test.cc",
"neteq/tools/neteq_performance_test.h",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
deps = [
":neteq",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_test_tools",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
":pcm16b",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../system_wrappers",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../test:test_support",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"//testing/gtest",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_source_set("neteq_quality_test_support") {
Refactor neteq_test_support. Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support. BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2252413002 Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 07:02:24 -07:00
testonly = true
sources = [
"neteq/tools/neteq_quality_test.cc",
"neteq/tools/neteq_quality_test.h",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
Refactor neteq_test_support. Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support. BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2252413002 Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 07:02:24 -07:00
}
deps = [
":neteq",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_test_tools",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:builtin_audio_decoder_factory",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../test:test_support",
Add gtest as a dependency for neteq_quality_test_support. Was removed in Patch Set 5 of https://codereview.webrtc.org/2252413002 but shouldn't have been, since it's actually required. https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h?l=17 BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2262173003 Cr-Commit-Position: refs/heads/master@{#13851}
2016-08-22 09:48:02 -07:00
"//testing/gtest",
Refactor neteq_test_support. Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support. BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2252413002 Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 07:02:24 -07:00
]
}
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
rtc_executable("rtp_encode") {
Migrated target RTPencode to GN TBR=henrik.lundin@webrtc.org BUG=6162 Review-Url: https://codereview.webrtc.org/2195353002 Cr-Commit-Position: refs/heads/master@{#13599}
2016-08-02 01:45:50 -07:00
testonly = true
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
deps = audio_coding_deps + [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
":audio_coding",
":neteq_input_audio_tools",
"../../api/audio_codecs/g711:audio_encoder_g711",
"../../api/audio_codecs/L16:audio_encoder_L16",
"../../api/audio_codecs/g722:audio_encoder_g722",
"../../api/audio_codecs/ilbc:audio_encoder_ilbc",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:system_wrappers_default",
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
"../../api/audio_codecs/isac:audio_encoder_isac",
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../rtc_base:rtc_base_approved",
]
Migrated target RTPencode to GN TBR=henrik.lundin@webrtc.org BUG=6162 Review-Url: https://codereview.webrtc.org/2195353002 Cr-Commit-Position: refs/heads/master@{#13599}
2016-08-02 01:45:50 -07:00
sources = [
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
"neteq/tools/rtp_encode.cc",
Migrated target RTPencode to GN TBR=henrik.lundin@webrtc.org BUG=6162 Review-Url: https://codereview.webrtc.org/2195353002 Cr-Commit-Position: refs/heads/master@{#13599}
2016-08-02 01:45:50 -07:00
]
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
defines = audio_coding_defines
Migrated target RTPencode to GN TBR=henrik.lundin@webrtc.org BUG=6162 Review-Url: https://codereview.webrtc.org/2195353002 Cr-Commit-Position: refs/heads/master@{#13599}
2016-08-02 01:45:50 -07:00
}
Replacing the legacy tool RTPjitter with a new rtp_jitter This new tool provides the similar functionality as the legacy tool, but is implemented using less legacy helpers. It also replaces RTPtimeshift and RTPchange. The most significant change versus the old RTPjitter tool is that the new tool takes the timing data in the form of integers in a text file (instead of the binary data file used by the old tool). This should make it easier to create custom timing files when needed. Bug: webrtc:2692 Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287 Reviewed-on: https://webrtc-review.googlesource.com/25700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:39:39 +01:00
rtc_executable("rtp_jitter") {
Migrated RTPchange Gyp target to GN. BUG=6162 Review-Url: https://codereview.webrtc.org/2194213002 Cr-Commit-Position: refs/heads/master@{#13601}
2016-08-02 02:05:03 -07:00
testonly = true
Replacing the legacy tool RTPjitter with a new rtp_jitter This new tool provides the similar functionality as the legacy tool, but is implemented using less legacy helpers. It also replaces RTPtimeshift and RTPchange. The most significant change versus the old RTPjitter tool is that the new tool takes the timing data in the form of integers in a text file (instead of the binary data file used by the old tool). This should make it easier to create custom timing files when needed. Bug: webrtc:2692 Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287 Reviewed-on: https://webrtc-review.googlesource.com/25700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:39:39 +01:00
deps = audio_coding_deps + [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:system_wrappers_default",
Replacing the legacy tool RTPjitter with a new rtp_jitter This new tool provides the similar functionality as the legacy tool, but is implemented using less legacy helpers. It also replaces RTPtimeshift and RTPchange. The most significant change versus the old RTPjitter tool is that the new tool takes the timing data in the form of integers in a text file (instead of the binary data file used by the old tool). This should make it easier to create custom timing files when needed. Bug: webrtc:2692 Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287 Reviewed-on: https://webrtc-review.googlesource.com/25700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:39:39 +01:00
"../rtp_rtcp:rtp_rtcp_format",
"../../api:array_view",
"../../rtc_base:rtc_base_approved",
]
Migrated RTPchange Gyp target to GN. BUG=6162 Review-Url: https://codereview.webrtc.org/2194213002 Cr-Commit-Position: refs/heads/master@{#13601}
2016-08-02 02:05:03 -07:00
sources = [
Replacing the legacy tool RTPjitter with a new rtp_jitter This new tool provides the similar functionality as the legacy tool, but is implemented using less legacy helpers. It also replaces RTPtimeshift and RTPchange. The most significant change versus the old RTPjitter tool is that the new tool takes the timing data in the form of integers in a text file (instead of the binary data file used by the old tool). This should make it easier to create custom timing files when needed. Bug: webrtc:2692 Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287 Reviewed-on: https://webrtc-review.googlesource.com/25700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:39:39 +01:00
"neteq/tools/rtp_jitter.cc",
Migrated RTPchange Gyp target to GN. BUG=6162 Review-Url: https://codereview.webrtc.org/2194213002 Cr-Commit-Position: refs/heads/master@{#13601}
2016-08-02 02:05:03 -07:00
]
Replacing the legacy tool RTPjitter with a new rtp_jitter This new tool provides the similar functionality as the legacy tool, but is implemented using less legacy helpers. It also replaces RTPtimeshift and RTPchange. The most significant change versus the old RTPjitter tool is that the new tool takes the timing data in the form of integers in a text file (instead of the binary data file used by the old tool). This should make it easier to create custom timing files when needed. Bug: webrtc:2692 Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287 Reviewed-on: https://webrtc-review.googlesource.com/25700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:39:39 +01:00
defines = audio_coding_defines
Migrated RTPchange Gyp target to GN. BUG=6162 Review-Url: https://codereview.webrtc.org/2194213002 Cr-Commit-Position: refs/heads/master@{#13601}
2016-08-02 02:05:03 -07:00
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("rtpcat") {
Migrated GN target rtpcat. BUG=6162 Review-Url: https://codereview.webrtc.org/2202193002 Cr-Commit-Position: refs/heads/master@{#13605}
2016-08-02 04:27:25 -07:00
testonly = true
sources = [
"neteq/tools/rtpcat.cc",
]
deps = [
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Migrated GN target rtpcat. BUG=6162 Review-Url: https://codereview.webrtc.org/2202193002 Cr-Commit-Position: refs/heads/master@{#13605}
2016-08-02 04:27:25 -07:00
"../../system_wrappers:system_wrappers_default",
"../../test:rtp_test_utils",
"//testing/gtest",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("rtp_analyze") {
GN migration of target audio_coding/neteq/rtp_analyze. Review-Url: https://codereview.webrtc.org/2185533003 Cr-Commit-Position: refs/heads/master@{#13533}
2016-07-26 06:46:19 -07:00
testonly = true
sources = [
"neteq/tools/rtp_analyze.cc",
]
deps = [
":neteq",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_test_tools",
GN migration of target audio_coding/neteq/rtp_analyze. Review-Url: https://codereview.webrtc.org/2185533003 Cr-Commit-Position: refs/heads/master@{#13533}
2016-07-26 06:46:19 -07:00
":pcm16b",
Replace remaining gflags usages with rtc_base/flags Continued from https://codereview.webrtc.org/2995363002 BUG=webrtc:7644 Review-Url: https://codereview.webrtc.org/3005483002 Cr-Commit-Position: refs/heads/master@{#19624}
2017-08-31 03:21:39 -07:00
"../../rtc_base:rtc_base_approved",
GN migration of target audio_coding/neteq/rtp_analyze. Review-Url: https://codereview.webrtc.org/2185533003 Cr-Commit-Position: refs/heads/master@{#13533}
2016-07-26 06:46:19 -07:00
"../../system_wrappers:system_wrappers_default",
"//testing/gtest",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN migration of target audio_coding/neteq/rtp_analyze. Review-Url: https://codereview.webrtc.org/2185533003 Cr-Commit-Position: refs/heads/master@{#13533}
2016-07-26 06:46:19 -07:00
}
}
Migrated GN target :neteq_opus_quality_test Migrated GN target :neteq_opus_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190, webrtc:2692 Review-Url: https://codereview.webrtc.org/2223933004 Cr-Commit-Position: refs/heads/master@{#13701}
2016-08-10 02:11:30 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("neteq_opus_quality_test") {
Migrated GN target :neteq_opus_quality_test Migrated GN target :neteq_opus_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190, webrtc:2692 Review-Url: https://codereview.webrtc.org/2223933004 Cr-Commit-Position: refs/heads/master@{#13701}
2016-08-10 02:11:30 -07:00
testonly = true
sources = [
"neteq/test/neteq_opus_quality_test.cc",
]
deps = [
":neteq",
Refactor neteq_test_support. Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support. BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2252413002 Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 07:02:24 -07:00
":neteq_quality_test_support",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_tools",
Migrated GN target :neteq_opus_quality_test Migrated GN target :neteq_opus_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190, webrtc:2692 Review-Url: https://codereview.webrtc.org/2223933004 Cr-Commit-Position: refs/heads/master@{#13701}
2016-08-10 02:11:30 -07:00
":webrtc_opus",
Replace gflags usages with rtc_base/flags in all targets based on test_main BUG=webrtc:7644 Review-Url: https://codereview.webrtc.org/2995363002 Cr-Commit-Position: refs/heads/master@{#19580}
2017-08-29 05:51:57 -07:00
"../../rtc_base:rtc_base_approved",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
Migrated GN target :neteq_opus_quality_test Migrated GN target :neteq_opus_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190, webrtc:2692 Review-Url: https://codereview.webrtc.org/2223933004 Cr-Commit-Position: refs/heads/master@{#13701}
2016-08-10 02:11:30 -07:00
"//testing/gtest",
]
}
Migrated GN target :neteq_speed_test Migrated GN target :neteq_speed_test from webrtc/modules/audio_coding/neteq/neteq.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2223343004 Cr-Commit-Position: refs/heads/master@{#13704}
2016-08-10 04:16:36 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("neteq_speed_test") {
Migrated GN target :neteq_speed_test Migrated GN target :neteq_speed_test from webrtc/modules/audio_coding/neteq/neteq.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2223343004 Cr-Commit-Position: refs/heads/master@{#13704}
2016-08-10 04:16:36 -07:00
testonly = true
sources = [
"neteq/test/neteq_speed_test.cc",
]
deps = [
":neteq",
":neteq_test_support",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Replace remaining gflags usages with rtc_base/flags Continued from https://codereview.webrtc.org/2995363002 BUG=webrtc:7644 Review-Url: https://codereview.webrtc.org/3005483002 Cr-Commit-Position: refs/heads/master@{#19624}
2017-08-31 03:21:39 -07:00
"../../rtc_base:rtc_base_approved",
Migrated GN target :neteq_speed_test Migrated GN target :neteq_speed_test from webrtc/modules/audio_coding/neteq/neteq.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2223343004 Cr-Commit-Position: refs/heads/master@{#13704}
2016-08-10 04:16:36 -07:00
"../../system_wrappers:system_wrappers_default",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Migrated GN target :neteq_speed_test Migrated GN target :neteq_speed_test from webrtc/modules/audio_coding/neteq/neteq.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2223343004 Cr-Commit-Position: refs/heads/master@{#13704}
2016-08-10 04:16:36 -07:00
"../../test:test_support",
Migrated GN target :audio_classifier_test Migrated GN target :audio_classifier_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2225333002 Cr-Commit-Position: refs/heads/master@{#13706}
2016-08-10 04:41:14 -07:00
]
}
Migrated GN target :neteq_ilbc_quality_test Migrated GN target :neteq_ilbc_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2221193004 Cr-Commit-Position: refs/heads/master@{#13708}
2016-08-10 04:55:20 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("neteq_ilbc_quality_test") {
Migrated GN target :neteq_ilbc_quality_test Migrated GN target :neteq_ilbc_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2221193004 Cr-Commit-Position: refs/heads/master@{#13708}
2016-08-10 04:55:20 -07:00
testonly = true
sources = [
"neteq/test/neteq_ilbc_quality_test.cc",
]
deps = [
":ilbc",
":neteq",
Refactor neteq_test_support. Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support. BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2252413002 Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 07:02:24 -07:00
":neteq_quality_test_support",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_tools",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../..:webrtc_common",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Migrated GN target :neteq_ilbc_quality_test Migrated GN target :neteq_ilbc_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2221193004 Cr-Commit-Position: refs/heads/master@{#13708}
2016-08-10 04:55:20 -07:00
"../../system_wrappers:system_wrappers_default",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
Migrated GN target :neteq_ilbc_quality_test Migrated GN target :neteq_ilbc_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2221193004 Cr-Commit-Position: refs/heads/master@{#13708}
2016-08-10 04:55:20 -07:00
"//testing/gtest",
]
}
Migrated GN target :neteq_isac_quality_test Migrated GN target :neteq_isac_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2226253002 Cr-Commit-Position: refs/heads/master@{#13710}
2016-08-10 05:04:47 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("neteq_isac_quality_test") {
Migrated GN target :neteq_isac_quality_test Migrated GN target :neteq_isac_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2226253002 Cr-Commit-Position: refs/heads/master@{#13710}
2016-08-10 05:04:47 -07:00
testonly = true
sources = [
"neteq/test/neteq_isac_quality_test.cc",
]
deps = [
":isac_fix",
":neteq",
Refactor neteq_test_support. Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support. BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2252413002 Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 07:02:24 -07:00
":neteq_quality_test_support",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
Migrated GN target :neteq_isac_quality_test Migrated GN target :neteq_isac_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2226253002 Cr-Commit-Position: refs/heads/master@{#13710}
2016-08-10 05:04:47 -07:00
"//testing/gtest",
]
}
Migrated GN target :neteq_pcmu_quality_test Migrated GN target :neteq_pcmu_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2224313002 Cr-Commit-Position: refs/heads/master@{#13711}
2016-08-10 05:06:27 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("neteq_pcmu_quality_test") {
Migrated GN target :neteq_pcmu_quality_test Migrated GN target :neteq_pcmu_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2224313002 Cr-Commit-Position: refs/heads/master@{#13711}
2016-08-10 05:06:27 -07:00
testonly = true
sources = [
"neteq/test/neteq_pcmu_quality_test.cc",
]
deps = [
":g711",
":neteq",
Refactor neteq_test_support. Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support. BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2252413002 Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 07:02:24 -07:00
":neteq_quality_test_support",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
Migrated GN target :neteq_pcmu_quality_test Migrated GN target :neteq_pcmu_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2224313002 Cr-Commit-Position: refs/heads/master@{#13711}
2016-08-10 05:06:27 -07:00
"//testing/gtest",
]
}
Migrated GN target :isac_fix_test Migrated GN target :isac_fix_test from webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2247233002 Cr-Commit-Position: refs/heads/master@{#13770}
2016-08-16 01:47:16 -07:00
Add pcm16b quality test supporting 48khz. Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Bug: webrtc:8836 Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Reviewed-on: https://webrtc-review.googlesource.com/47400 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21878}
2018-02-02 17:21:46 +01:00
rtc_executable("neteq_pcm16b_quality_test") {
testonly = true
sources = [
"neteq/test/neteq_pcm16b_quality_test.cc",
]
deps = [
":neteq",
":neteq_quality_test_support",
":pcm16b",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Add pcm16b quality test supporting 48khz. Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Bug: webrtc:8836 Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Reviewed-on: https://webrtc-review.googlesource.com/47400 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21878}
2018-02-02 17:21:46 +01:00
"../../test:test_main",
"//testing/gtest",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("isac_fix_test") {
Migrated GN target :isac_fix_test Migrated GN target :isac_fix_test from webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2247233002 Cr-Commit-Position: refs/heads/master@{#13770}
2016-08-16 01:47:16 -07:00
testonly = true
sources = [
"codecs/isac/fix/test/kenny.cc",
]
deps = [
":isac_fix",
Reland "iOS: Save perf results under Documents/perf_result.json" This will require a manual roll to downstream projects, since the //test:perf_test target was introduced. This is a reland of 10a8e7a9b5261a7e3ce19900ba3511be3b5911f8 Original change's description: > iOS: Save perf results under Documents/perf_result.json > > TBR=henrika@webrtc.org > > Bug: webrtc:7156 > Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5 > Reviewed-on: https://webrtc-review.googlesource.com/29202 > Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21244} R=henrika@webrtc.org, phoglund@webrtc.org Bug: webrtc:7156 Change-Id: I85fc7bc5fce0894af90017b71b9952b61b523424 Reviewed-on: https://webrtc-review.googlesource.com/37643 Reviewed-by: Patrik Höglund <phoglund@google.com> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21518}
2018-01-05 15:34:09 +01:00
"../../test:perf_test",
Migrated GN target :isac_fix_test Migrated GN target :isac_fix_test from webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2247233002 Cr-Commit-Position: refs/heads/master@{#13770}
2016-08-16 01:47:16 -07:00
"../../test:test_support",
]
Make it possible to isolate isac_fix_test. Successful swarming run: https://chromium-swarm.appspot.com/task?id=38769e8dd5d3e110&refresh=10&show_raw=1 BUG=chromium:755660 NOTRY=True Review-Url: https://codereview.webrtc.org/3009203002 Cr-Commit-Position: refs/heads/master@{#19738}
2017-09-07 23:18:35 -07:00
data = [
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
"../../resources/speech_and_misc_wb.pcm",
Make it possible to isolate isac_fix_test. Successful swarming run: https://chromium-swarm.appspot.com/task?id=38769e8dd5d3e110&refresh=10&show_raw=1 BUG=chromium:755660 NOTRY=True Review-Url: https://codereview.webrtc.org/3009203002 Cr-Commit-Position: refs/heads/master@{#19738}
2017-09-07 23:18:35 -07:00
]
Migrated GN target :isac_fix_test Migrated GN target :isac_fix_test from webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2247233002 Cr-Commit-Position: refs/heads/master@{#13770}
2016-08-16 01:47:16 -07:00
if (is_win) {
cflags = [
# Disable warnings to enable Win64 build, issue 1323.
"/wd4267", # size_t to int truncation
]
}
}
Migrated GN target :g711_test Migrated GN target :g711_test from webrtc/modules/audio_coding/codecs/g711/g711.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2273623002 Cr-Commit-Position: refs/heads/master@{#13864}
2016-08-23 08:08:23 -07:00
GN: Fix windows clang errors. Attempt 2. BUG=webrtc:6255 NOTRY=True Review-Url: https://codereview.webrtc.org/2281513002 Cr-Commit-Position: refs/heads/master@{#13942}
2016-08-26 13:31:24 -07:00
config("isac_test_warnings_config") {
if (is_win && is_clang) {
cflags = [
# Disable warnings failing when compiling with Clang on Windows.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
"-Wno-format",
]
}
}
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
rtc_source_set("isac_test_util") {
testonly = true
sources = [
"codecs/isac/main/util/utility.c",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"codecs/isac/main/util/utility.h",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("isac_test") {
Added GN target for isac_test. BUG=webrtc:6191 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2267423002 Cr-Commit-Position: refs/heads/master@{#13884}
2016-08-24 02:25:57 -07:00
testonly = true
sources = [
"codecs/isac/main/test/simpleKenny.c",
]
include_dirs = [
"codecs/isac/main/include",
"codecs/isac/main/test",
"codecs/isac/main/util",
]
deps = [
":isac",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
":isac_test_util",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Added GN target for isac_test. BUG=webrtc:6191 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2267423002 Cr-Commit-Position: refs/heads/master@{#13884}
2016-08-24 02:25:57 -07:00
]
GN: Fix windows clang errors. Attempt 2. BUG=webrtc:6255 NOTRY=True Review-Url: https://codereview.webrtc.org/2281513002 Cr-Commit-Position: refs/heads/master@{#13942}
2016-08-26 13:31:24 -07:00
configs += [ ":isac_test_warnings_config" ]
Added GN target for isac_test. BUG=webrtc:6191 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2267423002 Cr-Commit-Position: refs/heads/master@{#13884}
2016-08-24 02:25:57 -07:00
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("g711_test") {
Migrated GN target :g711_test Migrated GN target :g711_test from webrtc/modules/audio_coding/codecs/g711/g711.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2273623002 Cr-Commit-Position: refs/heads/master@{#13864}
2016-08-23 08:08:23 -07:00
testonly = true
sources = [
"codecs/g711/test/testG711.cc",
]
deps = [
":g711",
]
}
Migrated GN target :g722_test Migrated GN target :g722_test from webrtc/modules/audio_coding/codecs/g722/g722.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2275463002 Cr-Commit-Position: refs/heads/master@{#13865}
2016-08-23 08:36:10 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("g722_test") {
Migrated GN target :g722_test Migrated GN target :g722_test from webrtc/modules/audio_coding/codecs/g722/g722.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2275463002 Cr-Commit-Position: refs/heads/master@{#13865}
2016-08-23 08:36:10 -07:00
testonly = true
sources = [
"codecs/g722/test/testG722.cc",
]
deps = [
":g722",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
Migrated GN target :g722_test Migrated GN target :g722_test from webrtc/modules/audio_coding/codecs/g722/g722.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2275463002 Cr-Commit-Position: refs/heads/master@{#13865}
2016-08-23 08:36:10 -07:00
"../..:webrtc_common",
]
}
Added GN target for webrtc_opus_fec_test. BUG=webrtc:6191 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2268213002 Cr-Commit-Position: refs/heads/master@{#13893}
2016-08-24 06:11:18 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("isac_api_test") {
Migrated ILBC and ISAC test targets for GN. Migrated GN targets ilbc_test, isac_api_test, isac_switch_samprate_test from webrtc/modules/audio_coding/codecs NOTRY=True NOPRESUBMIT=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2270403002 Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 04:09:19 -07:00
testonly = true
sources = [
"codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc",
]
deps = [
":isac",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
":isac_test_util",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Migrated ILBC and ISAC test targets for GN. Migrated GN targets ilbc_test, isac_api_test, isac_switch_samprate_test from webrtc/modules/audio_coding/codecs NOTRY=True NOPRESUBMIT=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2270403002 Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 04:09:19 -07:00
]
include_dirs = [
"codecs/isac/main/include",
"codecs/isac/main/test",
"codecs/isac/main/util",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("isac_switch_samprate_test") {
Migrated ILBC and ISAC test targets for GN. Migrated GN targets ilbc_test, isac_api_test, isac_switch_samprate_test from webrtc/modules/audio_coding/codecs NOTRY=True NOPRESUBMIT=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2270403002 Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 04:09:19 -07:00
testonly = true
sources = [
"codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc",
]
deps = [
":isac",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
":isac_test_util",
Using fully qualified #include paths in isac code. WebRTC internal code should always used include paths that starts from the root of the project and that clearly identify the header file. This allows 'gn check' to actually keep dependencies under control because 'gn check' cannot enforce anything if the include path is not fully qualified (starting from the root of the project). Bug: webrtc:8815 Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc Reviewed-on: https://webrtc-review.googlesource.com/46101 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21847}
2018-02-01 14:12:55 +01:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Migrated ILBC and ISAC test targets for GN. Migrated GN targets ilbc_test, isac_api_test, isac_switch_samprate_test from webrtc/modules/audio_coding/codecs NOTRY=True NOPRESUBMIT=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2270403002 Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 04:09:19 -07:00
]
include_dirs = [
"codecs/isac/main/include",
"codecs/isac/main/test",
"codecs/isac/main/util",
"../../common_audio/signal_processing/include",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("ilbc_test") {
Migrated ILBC and ISAC test targets for GN. Migrated GN targets ilbc_test, isac_api_test, isac_switch_samprate_test from webrtc/modules/audio_coding/codecs NOTRY=True NOPRESUBMIT=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2270403002 Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 04:09:19 -07:00
testonly = true
sources = [
"codecs/ilbc/test/iLBC_test.c",
]
deps = [
":ilbc",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("webrtc_opus_fec_test") {
Added GN target for webrtc_opus_fec_test. BUG=webrtc:6191 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2268213002 Cr-Commit-Position: refs/heads/master@{#13893}
2016-08-24 06:11:18 -07:00
testonly = true
sources = [
"codecs/opus/opus_fec_test.cc",
]
deps = [
":webrtc_opus",
"../../common_audio",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
Added GN target for webrtc_opus_fec_test. BUG=webrtc:6191 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2268213002 Cr-Commit-Position: refs/heads/master@{#13893}
2016-08-24 06:11:18 -07:00
"//testing/gtest",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
Added GN target for webrtc_opus_fec_test. BUG=webrtc:6191 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2268213002 Cr-Commit-Position: refs/heads/master@{#13893}
2016-08-24 06:11:18 -07:00
}
}
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
rtc_source_set("audio_coding_unittests") {
testonly = true
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
sources = [
"acm2/acm_receiver_unittest.cc",
"acm2/audio_coding_module_unittest.cc",
"acm2/call_statistics_unittest.cc",
"acm2/codec_manager_unittest.cc",
"acm2/rent_a_codec_unittest.cc",
"audio_network_adaptor/audio_network_adaptor_impl_unittest.cc",
"audio_network_adaptor/bitrate_controller_unittest.cc",
"audio_network_adaptor/channel_controller_unittest.cc",
"audio_network_adaptor/controller_manager_unittest.cc",
"audio_network_adaptor/dtx_controller_unittest.cc",
Reland of "Log audio network adapter decisions in event log." This was originally reviewed https://codereview.webrtc.org/2559953002/ It was reverted due to a bug in the original CL, see https://codereview.webrtc.org/2631703002/ This CL is to fix and reland. BUG=webrtc:6845 Review-Url: https://codereview.webrtc.org/2644863002 Cr-Commit-Position: refs/heads/master@{#16242}
2017-01-24 04:54:59 -08:00
"audio_network_adaptor/event_log_writer_unittest.cc",
Introduce FecControllerRplrBased 1. Rename FecController to FecControllerPlrBased. 2. Introduce FecControllerRplrBased - a version of FecController that makes its decision based on RPLR instead of PLR. BUG=webrtc:7058 Review-Url: https://codereview.webrtc.org/2672933003 Cr-Commit-Position: refs/heads/master@{#17373}
2017-03-24 04:12:56 -07:00
"audio_network_adaptor/fec_controller_plr_based_unittest.cc",
"audio_network_adaptor/fec_controller_rplr_based_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"audio_network_adaptor/frame_length_controller_unittest.cc",
Avoid code duplication between PLR/RPLR-based FecController BUG=webrtc:7058 Review-Url: https://codereview.webrtc.org/2688613003 Cr-Commit-Position: refs/heads/master@{#17437}
2017-03-29 03:16:58 -07:00
"audio_network_adaptor/util/threshold_curve_unittest.cc",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"codecs/builtin_audio_decoder_factory_unittest.cc",
Injectable audio encoders: BuiltinAudioEncoderFactory This CL contains all the changes made to audio_coding while making audio encoders injectable. Apart from some small changes to webrtcvoiceengine, nothing here is hooked up to the outside world. Those changes will be added to a follow-up CL. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2695243005 Cr-Commit-Position: refs/heads/master@{#17569}
2017-04-06 10:03:21 -07:00
"codecs/builtin_audio_encoder_factory_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"codecs/cng/audio_encoder_cng_unittest.cc",
"codecs/cng/cng_unittest.cc",
"codecs/ilbc/ilbc_unittest.cc",
"codecs/isac/fix/source/filterbanks_unittest.cc",
"codecs/isac/fix/source/filters_unittest.cc",
"codecs/isac/fix/source/lpc_masking_model_unittest.cc",
"codecs/isac/fix/source/transform_unittest.cc",
"codecs/isac/main/source/audio_encoder_isac_unittest.cc",
"codecs/isac/main/source/isac_unittest.cc",
"codecs/isac/unittest.cc",
"codecs/legacy_encoded_audio_frame_unittest.cc",
"codecs/opus/audio_encoder_opus_unittest.cc",
Implement Opus bandwidth adjustment behind a FieldTrial Bug: webrtc:8522 Change-Id: I3a32ebfecd27ff74b507c2cee9e16aab17153442 Reviewed-on: https://webrtc-review.googlesource.com/22210 Commit-Queue: Alejandro Luebs <aluebs@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20799}
2017-11-20 11:13:56 -08:00
"codecs/opus/opus_bandwidth_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"codecs/opus/opus_unittest.cc",
"codecs/red/audio_encoder_copy_red_unittest.cc",
"neteq/audio_multi_vector_unittest.cc",
"neteq/audio_vector_unittest.cc",
"neteq/background_noise_unittest.cc",
"neteq/buffer_level_filter_unittest.cc",
"neteq/comfort_noise_unittest.cc",
"neteq/decision_logic_unittest.cc",
"neteq/decoder_database_unittest.cc",
"neteq/delay_manager_unittest.cc",
"neteq/delay_peak_detector_unittest.cc",
"neteq/dsp_helper_unittest.cc",
"neteq/dtmf_buffer_unittest.cc",
"neteq/dtmf_tone_generator_unittest.cc",
"neteq/expand_unittest.cc",
"neteq/merge_unittest.cc",
"neteq/mock/mock_buffer_level_filter.h",
"neteq/mock/mock_decoder_database.h",
"neteq/mock/mock_delay_manager.h",
"neteq/mock/mock_delay_peak_detector.h",
"neteq/mock/mock_dtmf_buffer.h",
"neteq/mock/mock_dtmf_tone_generator.h",
"neteq/mock/mock_expand.h",
"neteq/mock/mock_external_decoder_pcm16b.h",
"neteq/mock/mock_packet_buffer.h",
"neteq/mock/mock_red_payload_splitter.h",
Implement packet discard rate in NetEq. BUG=webrtc:7903 Change-Id: I819c9362671ca0b02c602d53e4dc39afdd8ec465 Reviewed-on: https://chromium-review.googlesource.com/555311 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18899}
2017-07-05 11:17:40 +02:00
"neteq/mock/mock_statistics_calculator.h",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"neteq/nack_tracker_unittest.cc",
"neteq/neteq_external_decoder_unittest.cc",
"neteq/neteq_impl_unittest.cc",
"neteq/neteq_network_stats_unittest.cc",
"neteq/neteq_stereo_unittest.cc",
"neteq/neteq_unittest.cc",
"neteq/normal_unittest.cc",
"neteq/packet_buffer_unittest.cc",
"neteq/post_decode_vad_unittest.cc",
"neteq/random_vector_unittest.cc",
"neteq/red_payload_splitter_unittest.cc",
NetEq/Stats: Don't let concealed_samples decrease When NetEq performs a merge operation, it will usually have to correct the stats for number of concealment samples produced, sometimes with decreasing it. This does not make sense in the context of the stats spec, and stats-consuming applications may not be prepared for it. With this change, only positive corrections are allowed for the concealed_samples value. This will sometimes lead to a small positive bias, but it will be negligible over time. Bug: webrtc:8253 Change-Id: Ie9de311ab16401f1a4b435f6269725901b8cf561 Reviewed-on: https://webrtc-review.googlesource.com/1583 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19941}
2017-09-25 12:22:46 +02:00
"neteq/statistics_calculator_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"neteq/sync_buffer_unittest.cc",
"neteq/tick_timer_unittest.cc",
"neteq/time_stretch_unittest.cc",
"neteq/timestamp_scaler_unittest.cc",
"neteq/tools/input_audio_file_unittest.cc",
"neteq/tools/packet_unittest.cc",
]
deps = [
":acm_receive_test",
":acm_send_test",
":audio_coding",
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq In short, what I did was to * Remove acm_common_defs.h (the stuff in it was used only by acm_codec_database.cc). * Move audio_coding_module_typedefs.h to a new build target. * Move the NetEqDecoder enum (and the associated NetEqDecoderToSdpAudioFormat function) to a new file in a new build target. BUG=webrtc:7243, webrtc:7244 Review-Url: https://codereview.webrtc.org/2723253005 Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 06:16:28 -08:00
":audio_coding_module_typedefs",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":audio_format_conversion",
":audio_network_adaptor",
":cng",
":g711",
":ilbc",
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
":isac",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":isac_c",
":isac_fix",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
":legacy_encoded_audio_frame",
Add missing files to audio_coding Bug: webrtc:7650 Change-Id: I8d7d8c3998799404ec6283896883d195468cdfdc Reviewed-on: https://webrtc-review.googlesource.com/37622 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21502}
2018-01-05 11:46:26 +01:00
":mocks",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":neteq",
":neteq_test_support",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_test_tools",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":pcm16b",
":red",
":rent_a_codec",
":webrtc_opus",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"../..:typedefs",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../..:webrtc_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../../api/audio_codecs:builtin_audio_encoder_factory",
Implement Opus bandwidth adjustment behind a FieldTrial Bug: webrtc:8522 Change-Id: I3a32ebfecd27ff74b507c2cee9e16aab17153442 Reviewed-on: https://webrtc-review.googlesource.com/22210 Commit-Queue: Alejandro Luebs <aluebs@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20799}
2017-11-20 11:13:56 -08:00
"../../api/audio_codecs/opus:audio_decoder_opus",
Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix We've done this previously with the other audio encoders, but Opus had to wait until all external users had been updated. BUG=webrtc:7847 Change-Id: I70422d7b6c715f32a43bee88febcf6b6155e18b3 Reviewed-on: https://webrtc-review.googlesource.com/8000 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20424}
2017-10-25 09:57:40 +02:00
"../../api/audio_codecs/opus:audio_encoder_opus",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Fix Gn Untracked headers in webrtc/common_audio Fixed following headers in this CL =================================== src/webrtc/common_audio/vad/mock/mock_vad.h src/webrtc/common_audio/mocks/mock_smoothing_filter.h src/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h BUG=webrtc:7648 Review-Url: https://codereview.webrtc.org/3013063002 Cr-Original-Commit-Position: refs/heads/master@{#19824} Review-Url: https://codereview.webrtc.org/3013673002 Cr-Commit-Position: refs/heads/master@{#19852}
2017-09-15 03:51:34 -07:00
"../../common_audio:mock_common_audio",
Reland: Add mock_rtc_event_log.h. Bug: webrtc:7642 Change-Id: I3f97a8b603e34819e1563b335c1eeb51f89b11ac Reviewed-on: https://webrtc-review.googlesource.com/37081 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21480}
2018-01-03 09:08:20 +01:00
"../../logging:mocks",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../../logging:rtc_event_audio",
Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead RtcEventLogImpl no longer hard-codes the way encoding is done. It now relies on RtcEventEncoder for it. This gives two benefits: 1. We can decide between the current encoding and the new encoding (which is still WIP) without code duplication (no need for RtcEventLogImplNew). 2. Encoding is done only when the event needs to be written to a file. This both avoids unnecessary encoding of events which don't end up getting written to a file, as well as is useful for the new, delta-based encoding, which is stateful. BUG=webrtc:8111 Change-Id: I9517132e5f96b8059002a66fde8d42d3a678c3bb Reviewed-on: https://webrtc-review.googlesource.com/1365 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20118}
2017-10-03 16:11:34 +02:00
"../../logging:rtc_event_log_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_base_tests_utils",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:sanitizer",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
"../../system_wrappers:cpu_features_api",
Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ AudioDecoder and AudioDecoderFactory are in webrtc/api/ now, so move their mocks to someplace central where tests from all over WebRTC are allowed to #include them. BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2798063004 Cr-Commit-Position: refs/heads/master@{#17619}
2017-04-10 05:15:48 -07:00
"../../test:audio_codec_mocks",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:field_trial",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:rtp_test_utils",
"../../test:test_common",
"../../test:test_support",
"//testing/gtest",
]
defines = audio_coding_defines
if (rtc_enable_protobuf) {
Replacing WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP with WEBRTC_ENABLE_PROTOBUF. Bug: None Change-Id: I595b094e7fcb12723614df3197a40833932ba0a0 Reviewed-on: https://chromium-review.googlesource.com/533074 Reviewed-by: Michael T <tschumim@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18568}
2017-06-13 11:49:29 +02:00
defines += [ "WEBRTC_NETEQ_UNITTEST_BITEXACT" ]
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
deps += [
":ana_config_proto",
":neteq_unittest_proto",
]
}
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
# For backwards compatibility only! Use
# webrtc/api/audio_codecs:audio_codecs_api instead.
# TODO(kwiberg): Remove this.
rtc_source_set("audio_decoder_interface") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
sources = [
"codecs/audio_decoder.h",
]
deps = [
"../../api/audio_codecs:audio_codecs_api",
]
}
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
# For backwards compatibility only! Use
# webrtc/api/audio_codecs:audio_codecs_api instead.
# TODO(ossu): Remove this.
rtc_source_set("audio_encoder_interface") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
sources = [
"codecs/audio_encoder.h",
]
deps = [
"../../api/audio_codecs:audio_codecs_api",
]
}
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