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webrtc_m130/webrtc/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")
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
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" ]
}
if (!build_with_mozilla) {
if (current_cpu == "arm") {
audio_codec_deps += [ ":isac_fix" ]
} else {
audio_codec_deps += [ ":isac" ]
}
audio_codec_deps += [ ":g722" ]
}
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") {
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 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",
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
"../../base:rtc_base_approved",
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
]
}
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
rtc_source_set("audio_encoder_factory_interface") {
sources = [
"codecs/audio_encoder_factory.h",
]
deps = [
":audio_encoder_interface",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
]
}
rtc_static_library("builtin_audio_encoder_factory") {
# TODO(kjellander): Remove (bugs.webrtc.org/6828)
# Errors on cyclic dependency with :isac_fix if enabled.
check_includes = false
sources = [
"codecs/builtin_audio_encoder_factory.cc",
"codecs/builtin_audio_encoder_factory.h",
]
deps = [
"../..:webrtc_common",
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
"../../base:protobuf_utils",
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
"../../base:rtc_base_approved",
":audio_encoder_factory_interface",
] + audio_codec_deps
defines = audio_codec_defines
}
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("builtin_audio_decoder_factory_internal") {
New interface (AudioDecoderFactory), with an implementation This is a first draft of what we're hoping to use to create all AudioDecoder instances. Follow-up CLs will start using this internally in NetEq instead of calling constructors manually. BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/1917163002 Cr-Commit-Position: refs/heads/master@{#12548}
2016-04-28 14:23:32 -07:00
sources = [
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_internal.cc",
"codecs/builtin_audio_decoder_factory_internal.h",
New interface (AudioDecoderFactory), with an implementation This is a first draft of what we're hoping to use to create all AudioDecoder instances. Follow-up CLs will start using this internally in NetEq instead of calling constructors manually. BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/1917163002 Cr-Commit-Position: refs/heads/master@{#12548}
2016-04-28 14:23:32 -07:00
]
deps = [
"../..:webrtc_common",
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
"../../base:rtc_base_approved",
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",
New interface (AudioDecoderFactory), with an implementation This is a first draft of what we're hoping to use to create all AudioDecoder instances. Follow-up CLs will start using this internally in NetEq instead of calling constructors manually. BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/1917163002 Cr-Commit-Position: refs/heads/master@{#12548}
2016-04-28 14:23:32 -07:00
] + audio_codec_deps
defines = audio_codec_defines
}
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") {
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 = [
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",
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
"../../base:protobuf_utils",
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
"../../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",
":audio_encoder_interface",
":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",
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") {
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 + [
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 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",
":audio_encoder_interface",
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",
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
"../../base:rtc_base_approved",
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 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",
Add proper dependencies for webrtc/modules/audio_coding/codecs The audio_decoder_interface and audio_encoder_interface targets are depending on rtc_base_approved headers but don't declare such dependencies. NOTRY=True Review URL: https://codereview.webrtc.org/1916953002 Cr-Commit-Position: refs/heads/master@{#12500}
2016-04-25 21:59:50 -07:00
"../../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: 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_encoder_interface") {
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/audio_encoder.cc",
"codecs/audio_encoder.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
deps = [
"../..:webrtc_common",
Add proper dependencies for webrtc/modules/audio_coding/codecs The audio_decoder_interface and audio_encoder_interface targets are depending on rtc_base_approved headers but don't declare such dependencies. NOTRY=True Review URL: https://codereview.webrtc.org/1916953002 Cr-Commit-Position: refs/heads/master@{#12500}
2016-04-25 21:59:50 -07:00
"../../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
]
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("cng_config") {
include_dirs = [
Reland rev 7041 with BUILD.gn files. Original description: Audio codecs to include webrtc/typedefs.h Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h CL Generated with: $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g" BUG=3777 R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:28:48 +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
"codecs/cng/include",
]
}
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") {
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 = [
":audio_encoder_interface",
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",
"../../base:rtc_base_approved",
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",
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") {
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 = [
":audio_encoder_interface",
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
"../../base:rtc_base_approved",
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",
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") {
include_dirs = [
Reland rev 7041 with BUILD.gn files. Original description: Audio codecs to include webrtc/typedefs.h Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h CL Generated with: $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g" BUG=3777 R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:28:48 +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
"codecs/g711/include",
]
}
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") {
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 = [
":audio_encoder_interface",
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",
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
"../../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") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
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 = [
"../..: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") {
include_dirs = [
Reland rev 7041 with BUILD.gn files. Original description: Audio codecs to include webrtc/typedefs.h Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h CL Generated with: $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g" BUG=3777 R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:28:48 +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
"codecs/g722/include",
]
}
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") {
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 = [
":audio_encoder_interface",
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",
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
"../../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") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
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 = [
"../..: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") {
include_dirs = [
Reland rev 7041 with BUILD.gn files. Original description: Audio codecs to include webrtc/typedefs.h Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h CL Generated with: $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g" BUG=3777 R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:28:48 +00:00
"../../..",
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/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") {
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 = [
":audio_encoder_interface",
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",
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
"../../base:rtc_base_approved",
"../../common_audio",
]
public_deps = [
":ilbc_c",
]
}
rtc_source_set("ilbc_c") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
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 = [
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_encoder_interface",
"../..: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
"../../base:rtc_base_approved",
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",
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 = [
"codecs/isac/audio_encoder_isac_t.h",
"codecs/isac/audio_encoder_isac_t_impl.h",
"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 = [
":audio_encoder_interface",
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",
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",
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
"../../base:rtc_base_approved",
]
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") {
include_dirs = [
Reland rev 7041 with BUILD.gn files. Original description: Audio codecs to include webrtc/typedefs.h Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h CL Generated with: $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g" BUG=3777 R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:28:48 +00:00
"../../..",
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",
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") {
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 = [
":audio_encoder_interface",
":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") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
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",
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",
"../../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
"../../common_audio",
]
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") {
include_dirs = [
Reland rev 7041 with BUILD.gn files. Original description: Audio codecs to include webrtc/typedefs.h Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h CL Generated with: $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g" BUG=3777 R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:28:48 +00:00
"../../..",
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",
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") {
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 = [
":audio_encoder_interface",
":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") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
sources = [
"codecs/isac/fix/source/codec.h",
"codecs/isac/fix/source/fft.c",
"codecs/isac/fix/source/fft.h",
"codecs/isac/fix/source/settings.h",
]
public_configs = [ ":isac_fix_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
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") {
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
visibility = [ ":*" ] # Only targets in this file can depend on this.
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/entropy_coding.h",
"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/structs.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 = [
":audio_encoder_interface",
":isac_common",
"../..: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
"../../base:rtc_base_approved",
"../../common_audio",
"../../system_wrappers",
]
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) {
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/lattice_armv7.S",
"codecs/isac/fix/source/pitch_filter_armv6.S",
]
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_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
}
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",
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
"../../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
"../../common_audio",
]
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") {
include_dirs = [
Reland rev 7041 with BUILD.gn files. Original description: Audio codecs to include webrtc/typedefs.h Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h CL Generated with: $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g" BUG=3777 R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:28:48 +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
"codecs/pcm16b/include",
]
}
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") {
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",
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 = [
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
":audio_encoder_interface",
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
":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",
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
"../../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") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
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 = [
"../..: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("opus_config") {
Reland rev 7041 with BUILD.gn files. Original description: Audio codecs to include webrtc/typedefs.h Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h CL Generated with: $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g" BUG=3777 R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:28:48 +00:00
include_dirs = [ "../../.." ]
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") {
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 = [
":audio_encoder_interface",
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 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",
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
"../../base:protobuf_utils",
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
"../../base:rtc_base_approved",
Rename base/analytics/ to base/numerics/ BUG=webrtc:6832 Review-Url: https://codereview.webrtc.org/2626203002 Cr-Commit-Position: refs/heads/master@{#16060}
2017-01-13 09:14:33 -08:00
"../../base:rtc_numerics",
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",
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",
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",
]
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
Opus: Move complexity variable out of conditional build flag BUG=webrtc:6708 Review-Url: https://codereview.webrtc.org/2535933002 Cr-Commit-Position: refs/heads/master@{#15277}
2016-11-28 11:15:54 -08:00
if (rtc_opus_variable_complexity) {
defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
} else {
defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ]
}
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) {
include_dirs = [ getenv("DIST") + "/include/opus" ]
}
}
rtc_source_set("webrtc_opus_c") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
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 = [
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",
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
"../../base:rtc_base_approved",
]
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") {
sources = [
"audio_network_adaptor/debug_dump.proto",
]
Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ ) Reason for revert: Breaks chromium.webrtc.fyi. Original issue's description: > Log audio network adapter decisions in event log. > > BUG=webrtc:6845 > > Review-Url: https://codereview.webrtc.org/2559953002 > Cr-Commit-Position: refs/heads/master@{#16053} > Committed: https://chromium.googlesource.com/external/webrtc/+/3663681b5d05682f38e75d9eaf6049299f62fb02 TBR=minyue@webrtc.org,henrik.lundin@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6845 Review-Url: https://codereview.webrtc.org/2631703002 Cr-Commit-Position: refs/heads/master@{#16054}
2017-01-13 06:52:12 -08:00
proto_out_dir = "webrtc/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",
]
proto_out_dir = "webrtc/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
}
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") {
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 = [
"audio_network_adaptor/audio_network_adaptor.cc",
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
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",
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
"../../base:protobuf_utils",
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
"../../base:rtc_base_approved",
"../../common_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",
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",
]
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",
]
defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
}
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 = [
"../../api/audio_codecs:audio_codecs_api",
"../../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
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") {
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_decoder_impl.cc",
"neteq/audio_decoder_impl.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",
":g711",
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_fix",
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",
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
":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",
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",
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
"../../base:gtest_prod",
"../../base:rtc_base_approved",
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",
"../../system_wrappers",
]
Making sure neteq gets compiled with OPUS. All WebRTC calls on GN were failing because we failed to add OPUS as a receive codec. The reason was that the WEBRTC_CODEC_OPUS define wasn't set for the audio_decoder_impl.cc file; this CL fixes that. BUG=None R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42309004 Cr-Commit-Position: refs/heads/master@{#8672} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8672 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 16:23:24 +00:00
defines = []
Add a gyp/gn variable for whether to use iLBC or not BUG=webrtc:5415 Review URL: https://codereview.webrtc.org/1578953003 Cr-Commit-Position: refs/heads/master@{#11291}
2016-01-18 06:38:32 -08:00
if (rtc_include_ilbc) {
defines += [ "WEBRTC_CODEC_ILBC" ]
deps += [ ":ilbc" ]
}
Making sure neteq gets compiled with OPUS. All WebRTC calls on GN were failing because we failed to add OPUS as a receive codec. The reason was that the WEBRTC_CODEC_OPUS define wasn't set for the audio_decoder_impl.cc file; this CL fixes that. BUG=None R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42309004 Cr-Commit-Position: refs/heads/master@{#8672} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8672 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 16:23:24 +00:00
if (rtc_include_opus) {
defines += [ "WEBRTC_CODEC_OPUS" ]
deps += [ ":webrtc_opus" ]
}
Don't link with audio codecs that we don't use We used to link with all audio codecs unconditionally (except Opus); this patch makes gyp and gn only link to the ones that are used. This unfortunately fails to have a measurable impact on Chromium binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC fix were already being excluded from Chromium by some other means, likely just the linker omitting compilation units with no incoming references. (This was previously landed as revisions 10046 and 10060, and got reverted because it broke several of the Chromium FYI bots.) BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1368843003 Cr-Commit-Position: refs/heads/master@{#10127}
2015-09-30 21:54:21 -07:00
if (!build_with_mozilla) {
if (current_cpu == "arm") {
defines += [ "WEBRTC_CODEC_ISACFX" ]
deps += [ ":isac_fix" ]
} else {
defines += [ "WEBRTC_CODEC_ISAC" ]
deps += [ ":isac" ]
}
defines += [ "WEBRTC_CODEC_G722" ]
deps += [ ":g722" ]
}
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") {
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 = [
":audio_encoder_interface",
":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",
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",
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
"../../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
}
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) {
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") {
testonly = true
public_deps = [
":RTPchange",
":RTPencode",
":RTPjitter",
":RTPtimeshift",
":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",
":neteq_pcmu_quality_test",
":neteq_speed_test",
":rtp_analyze",
":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
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
# Skip restricting visibility on mobile platforms since the tests on those
# gets additional generated targets which would require many lines here to
# cover (which would be confusing to read and hard to maintain).
if (!is_android && !is_ios) {
visibility = [ "//webrtc/modules:modules_tests" ]
}
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 = [
"test/APITest.cc",
"test/Channel.cc",
"test/EncodeDecodeTest.cc",
"test/PCMFile.cc",
"test/PacketLossTest.cc",
"test/RTPFile.cc",
"test/TestAllCodecs.cc",
"test/TestRedFec.cc",
"test/TestStereo.cc",
"test/TestVADDTX.cc",
"test/Tester.cc",
"test/TwoWayCommunication.cc",
"test/iSACTest.cc",
"test/opus_test.cc",
"test/target_delay_unittest.cc",
"test/utility.cc",
]
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",
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 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",
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
"../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../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
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
# Skip restricting visibility on mobile platforms since the tests on those
# gets additional generated targets which would require many lines here to
# cover (which would be confusing to read and hard to maintain).
if (!is_android && !is_ios) {
visibility = [ "//webrtc:webrtc_perf_tests" ]
}
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",
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_opus",
"../..:webrtc_common",
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
"../../base:protobuf_utils",
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
"../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../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",
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
"../../base:rtc_base_approved",
"../../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 + [
":audio_coding",
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
":audio_encoder_interface",
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
"../../base:rtc_base_approved",
"../../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",
"test/PCMFile.cc",
"test/delay_test.cc",
"test/utility.cc",
]
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",
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",
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
"../../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",
"../../test:test_support",
"../rtp_rtcp",
"//testing/gtest",
"//third_party/gflags:gflags",
]
} # 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",
"test/PCMFile.cc",
"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",
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",
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
"../../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",
"../../test:test_support",
"../rtp_rtcp",
"//testing/gtest",
"//third_party/gflags:gflags",
]
} # 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 =
[ "//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
}
deps = []
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",
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
"../../base:protobuf_utils",
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",
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",
]
Restructure protobuf targets to fit with downstream requirements. The paths of the protobuf output files needs to match the actual tree path in order for Bazel builds to work. BUG=webrtc:6412 NOTRY=True Review-Url: https://codereview.webrtc.org/2716413004 Cr-Commit-Position: refs/heads/master@{#16912}
2017-02-28 08:01:46 -08:00
proto_out_dir = "webrtc/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
Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ ) Reason for revert: It causes a Chromium build error: ERROR at //third_party/webrtc/test/BUILD.gn:113:5: Can't load input file. "//third_party/gflags", Original issue's description: > Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) > > Reason for revert: > Try to fix the webrtc/test/fuzzers issue and reland this CL because it > contains lots of fixes for our BUILD.gn files. > > Original issue's description: > > Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ ) > > > > Reason for revert: > > Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6). > > > > Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer. > > > > Original issue's description: > > > Enable GN check for webrtc/base > > > > > > It's not possible to enable it for the rtc_base_approved > > > target but since a larger refactoring is ongoing for webrtc/base > > > this CL doesn't attempt to fix that. > > > > > > Changes made: > > > * Move webrtc/system_wrappers/include/stringize_macros.h into > > > webrtc/base:rtc_base_approved_unittests (and corresponding > > > unit test to rtc_base_approved_unittests). > > > * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target. > > > * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into > > > webrtc/base. > > > * Remove unused use include of webrtc/base/fileutils.h in > > > webrtc/base/pathutils.cc > > > > > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > > NOTRY=True > > > > > > Review-Url: https://codereview.webrtc.org/2717083002 > > > Cr-Commit-Position: refs/heads/master@{#17766} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/ed754e71ae8866db641677073274e86fe704eeac > > > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838683002 > > Cr-Commit-Position: refs/heads/master@{#17849} > > Committed: https://chromium.googlesource.com/external/webrtc/+/11ed366c487a938815cd52ad2ab5467b0f90e1ae > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > Review-Url: https://codereview.webrtc.org/2840453004 > Cr-Commit-Position: refs/heads/master@{#17876} > Committed: https://chromium.googlesource.com/external/webrtc/+/7054085e59c3123613cd0014bedb0fe91a56e26f TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828, webrtc:3806, webrtc:7480 Review-Url: https://codereview.webrtc.org/2846483002 Cr-Commit-Position: refs/heads/master@{#17877}
2017-04-26 00:38:48 -07:00
rtc_static_library("rtc_event_log_source") {
testonly = true
# TODO(kjellander): Remove (bugs.webrtc.org/6828)
# Needs call.h to be moved to webrtc/api first.
check_includes = false
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 = [
"../../base:rtc_base_approved",
"../../logging:rtc_event_log_parser",
]
public_deps = [
"../../logging:rtc_event_log_proto",
]
}
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",
]
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",
"../../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",
"//third_party/gflags",
]
}
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: 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 = []
deps = []
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
}
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
}
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",
"../../base:rtc_base_approved",
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",
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",
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",
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
"../../base:rtc_base_approved",
"../../system_wrappers",
"../../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",
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",
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
"../../base:rtc_base_approved",
"../../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
"//third_party/gflags",
]
}
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
config("neteq_tools_config") {
Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ ) Reason for revert: It causes a Chromium build error: ERROR at //third_party/webrtc/test/BUILD.gn:113:5: Can't load input file. "//third_party/gflags", Original issue's description: > Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) > > Reason for revert: > Try to fix the webrtc/test/fuzzers issue and reland this CL because it > contains lots of fixes for our BUILD.gn files. > > Original issue's description: > > Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ ) > > > > Reason for revert: > > Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6). > > > > Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer. > > > > Original issue's description: > > > Enable GN check for webrtc/base > > > > > > It's not possible to enable it for the rtc_base_approved > > > target but since a larger refactoring is ongoing for webrtc/base > > > this CL doesn't attempt to fix that. > > > > > > Changes made: > > > * Move webrtc/system_wrappers/include/stringize_macros.h into > > > webrtc/base:rtc_base_approved_unittests (and corresponding > > > unit test to rtc_base_approved_unittests). > > > * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target. > > > * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into > > > webrtc/base. > > > * Remove unused use include of webrtc/base/fileutils.h in > > > webrtc/base/pathutils.cc > > > > > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > > NOTRY=True > > > > > > Review-Url: https://codereview.webrtc.org/2717083002 > > > Cr-Commit-Position: refs/heads/master@{#17766} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/ed754e71ae8866db641677073274e86fe704eeac > > > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838683002 > > Cr-Commit-Position: refs/heads/master@{#17849} > > Committed: https://chromium.googlesource.com/external/webrtc/+/11ed366c487a938815cd52ad2ab5467b0f90e1ae > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > Review-Url: https://codereview.webrtc.org/2840453004 > Cr-Commit-Position: refs/heads/master@{#17876} > Committed: https://chromium.googlesource.com/external/webrtc/+/7054085e59c3123613cd0014bedb0fe91a56e26f TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828, webrtc:3806, webrtc:7480 Review-Url: https://codereview.webrtc.org/2846483002 Cr-Commit-Position: refs/heads/master@{#17877}
2017-04-26 00:38:48 -07:00
include_dirs = [ "tools" ]
}
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") {
Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ ) Reason for revert: It causes a Chromium build error: ERROR at //third_party/webrtc/test/BUILD.gn:113:5: Can't load input file. "//third_party/gflags", Original issue's description: > Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) > > Reason for revert: > Try to fix the webrtc/test/fuzzers issue and reland this CL because it > contains lots of fixes for our BUILD.gn files. > > Original issue's description: > > Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ ) > > > > Reason for revert: > > Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6). > > > > Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer. > > > > Original issue's description: > > > Enable GN check for webrtc/base > > > > > > It's not possible to enable it for the rtc_base_approved > > > target but since a larger refactoring is ongoing for webrtc/base > > > this CL doesn't attempt to fix that. > > > > > > Changes made: > > > * Move webrtc/system_wrappers/include/stringize_macros.h into > > > webrtc/base:rtc_base_approved_unittests (and corresponding > > > unit test to rtc_base_approved_unittests). > > > * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target. > > > * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into > > > webrtc/base. > > > * Remove unused use include of webrtc/base/fileutils.h in > > > webrtc/base/pathutils.cc > > > > > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > > NOTRY=True > > > > > > Review-Url: https://codereview.webrtc.org/2717083002 > > > Cr-Commit-Position: refs/heads/master@{#17766} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/ed754e71ae8866db641677073274e86fe704eeac > > > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838683002 > > Cr-Commit-Position: refs/heads/master@{#17849} > > Committed: https://chromium.googlesource.com/external/webrtc/+/11ed366c487a938815cd52ad2ab5467b0f90e1ae > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > Review-Url: https://codereview.webrtc.org/2840453004 > Cr-Commit-Position: refs/heads/master@{#17876} > Committed: https://chromium.googlesource.com/external/webrtc/+/7054085e59c3123613cd0014bedb0fe91a56e26f TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828, webrtc:3806, webrtc:7480 Review-Url: https://codereview.webrtc.org/2846483002 Cr-Commit-Position: refs/heads/master@{#17877}
2017-04-26 00:38:48 -07:00
sources = [
"neteq/tools/fake_decode_from_file.cc",
"neteq/tools/fake_decode_from_file.h",
"neteq/tools/input_audio_file.cc",
"neteq/tools/input_audio_file.h",
"neteq/tools/neteq_replacement_input.cc",
"neteq/tools/neteq_replacement_input.h",
"neteq/tools/resample_input_audio_file.cc",
"neteq/tools/resample_input_audio_file.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
]
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 = [
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
"../../common_audio",
"../rtp_rtcp",
]
public_deps = [
":neteq_tools_minimal",
]
}
rtc_source_set("neteq_test_tools") {
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",
Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ ) Reason for revert: It causes a Chromium build error: ERROR at //third_party/webrtc/test/BUILD.gn:113:5: Can't load input file. "//third_party/gflags", Original issue's description: > Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) > > Reason for revert: > Try to fix the webrtc/test/fuzzers issue and reland this CL because it > contains lots of fixes for our BUILD.gn files. > > Original issue's description: > > Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ ) > > > > Reason for revert: > > Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6). > > > > Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer. > > > > Original issue's description: > > > Enable GN check for webrtc/base > > > > > > It's not possible to enable it for the rtc_base_approved > > > target but since a larger refactoring is ongoing for webrtc/base > > > this CL doesn't attempt to fix that. > > > > > > Changes made: > > > * Move webrtc/system_wrappers/include/stringize_macros.h into > > > webrtc/base:rtc_base_approved_unittests (and corresponding > > > unit test to rtc_base_approved_unittests). > > > * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target. > > > * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into > > > webrtc/base. > > > * Remove unused use include of webrtc/base/fileutils.h in > > > webrtc/base/pathutils.cc > > > > > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > > NOTRY=True > > > > > > Review-Url: https://codereview.webrtc.org/2717083002 > > > Cr-Commit-Position: refs/heads/master@{#17766} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/ed754e71ae8866db641677073274e86fe704eeac > > > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838683002 > > Cr-Commit-Position: refs/heads/master@{#17849} > > Committed: https://chromium.googlesource.com/external/webrtc/+/11ed366c487a938815cd52ad2ab5467b0f90e1ae > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > Review-Url: https://codereview.webrtc.org/2840453004 > Cr-Commit-Position: refs/heads/master@{#17876} > Committed: https://chromium.googlesource.com/external/webrtc/+/7054085e59c3123613cd0014bedb0fe91a56e26f TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828, webrtc:3806, webrtc:7480 Review-Url: https://codereview.webrtc.org/2846483002 Cr-Commit-Position: refs/heads/master@{#17877}
2017-04-26 00:38:48 -07:00
"neteq/tools/rtp_file_source.cc",
"neteq/tools/rtp_file_source.h",
"neteq/tools/rtp_generator.cc",
"neteq/tools/rtp_generator.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
public_configs = [ ":neteq_tools_config" ]
Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ ) Reason for revert: It causes a Chromium build error: ERROR at //third_party/webrtc/test/BUILD.gn:113:5: Can't load input file. "//third_party/gflags", Original issue's description: > Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) > > Reason for revert: > Try to fix the webrtc/test/fuzzers issue and reland this CL because it > contains lots of fixes for our BUILD.gn files. > > Original issue's description: > > Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ ) > > > > Reason for revert: > > Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6). > > > > Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer. > > > > Original issue's description: > > > Enable GN check for webrtc/base > > > > > > It's not possible to enable it for the rtc_base_approved > > > target but since a larger refactoring is ongoing for webrtc/base > > > this CL doesn't attempt to fix that. > > > > > > Changes made: > > > * Move webrtc/system_wrappers/include/stringize_macros.h into > > > webrtc/base:rtc_base_approved_unittests (and corresponding > > > unit test to rtc_base_approved_unittests). > > > * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target. > > > * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into > > > webrtc/base. > > > * Remove unused use include of webrtc/base/fileutils.h in > > > webrtc/base/pathutils.cc > > > > > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > > NOTRY=True > > > > > > Review-Url: https://codereview.webrtc.org/2717083002 > > > Cr-Commit-Position: refs/heads/master@{#17766} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/ed754e71ae8866db641677073274e86fe704eeac > > > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838683002 > > Cr-Commit-Position: refs/heads/master@{#17849} > > Committed: https://chromium.googlesource.com/external/webrtc/+/11ed366c487a938815cd52ad2ab5467b0f90e1ae > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > Review-Url: https://codereview.webrtc.org/2840453004 > Cr-Commit-Position: refs/heads/master@{#17876} > Committed: https://chromium.googlesource.com/external/webrtc/+/7054085e59c3123613cd0014bedb0fe91a56e26f TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828, webrtc:3806, webrtc:7480 Review-Url: https://codereview.webrtc.org/2846483002 Cr-Commit-Position: refs/heads/master@{#17877}
2017-04-26 00:38:48 -07:00
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" ]
}
deps = [
":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",
Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ ) Reason for revert: It causes a Chromium build error: ERROR at //third_party/webrtc/test/BUILD.gn:113:5: Can't load input file. "//third_party/gflags", Original issue's description: > Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) > > Reason for revert: > Try to fix the webrtc/test/fuzzers issue and reland this CL because it > contains lots of fixes for our BUILD.gn files. > > Original issue's description: > > Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ ) > > > > Reason for revert: > > Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6). > > > > Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer. > > > > Original issue's description: > > > Enable GN check for webrtc/base > > > > > > It's not possible to enable it for the rtc_base_approved > > > target but since a larger refactoring is ongoing for webrtc/base > > > this CL doesn't attempt to fix that. > > > > > > Changes made: > > > * Move webrtc/system_wrappers/include/stringize_macros.h into > > > webrtc/base:rtc_base_approved_unittests (and corresponding > > > unit test to rtc_base_approved_unittests). > > > * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target. > > > * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into > > > webrtc/base. > > > * Remove unused use include of webrtc/base/fileutils.h in > > > webrtc/base/pathutils.cc > > > > > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > > NOTRY=True > > > > > > Review-Url: https://codereview.webrtc.org/2717083002 > > > Cr-Commit-Position: refs/heads/master@{#17766} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/ed754e71ae8866db641677073274e86fe704eeac > > > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838683002 > > Cr-Commit-Position: refs/heads/master@{#17849} > > Committed: https://chromium.googlesource.com/external/webrtc/+/11ed366c487a938815cd52ad2ab5467b0f90e1ae > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > Review-Url: https://codereview.webrtc.org/2840453004 > Cr-Commit-Position: refs/heads/master@{#17876} > Committed: https://chromium.googlesource.com/external/webrtc/+/7054085e59c3123613cd0014bedb0fe91a56e26f TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828, webrtc:3806, webrtc:7480 Review-Url: https://codereview.webrtc.org/2846483002 Cr-Commit-Position: refs/heads/master@{#17877}
2017-04-26 00:38:48 -07:00
"../..:webrtc_common",
"../../base:rtc_base_approved",
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) Reason for revert: Fourth attempt to land. Waiting for https://codereview.webrtc.org/2845013003 to avoid conflicts on webrtc/modules/audio_coding:neteq_unittest_tools. Original issue's description: > Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ ) > > Reason for revert: > Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6). > > Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer. > > Original issue's description: > > Enable GN check for webrtc/base > > > > It's not possible to enable it for the rtc_base_approved > > target but since a larger refactoring is ongoing for webrtc/base > > this CL doesn't attempt to fix that. > > > > Changes made: > > * Move webrtc/system_wrappers/include/stringize_macros.h into > > webrtc/base:rtc_base_approved_unittests (and corresponding > > unit test to rtc_base_approved_unittests). > > * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target. > > * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into > > webrtc/base. > > * Remove unused use include of webrtc/base/fileutils.h in > > webrtc/base/pathutils.cc > > > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2717083002 > > Cr-Commit-Position: refs/heads/master@{#17766} > > Committed: https://chromium.googlesource.com/external/webrtc/+/ed754e71ae8866db641677073274e86fe704eeac > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > NOTRY=True > > Review-Url: https://codereview.webrtc.org/2838683002 > Cr-Commit-Position: refs/heads/master@{#17849} > Committed: https://chromium.googlesource.com/external/webrtc/+/11ed366c487a938815cd52ad2ab5467b0f90e1ae TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6828, webrtc:3806, webrtc:7480 Review-Url: https://codereview.webrtc.org/2852663002 Cr-Commit-Position: refs/heads/master@{#17927}
2017-04-28 05:24:50 -07:00
"../../base:rtc_base_tests_utils",
Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ ) Reason for revert: It causes a Chromium build error: ERROR at //third_party/webrtc/test/BUILD.gn:113:5: Can't load input file. "//third_party/gflags", Original issue's description: > Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) > > Reason for revert: > Try to fix the webrtc/test/fuzzers issue and reland this CL because it > contains lots of fixes for our BUILD.gn files. > > Original issue's description: > > Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ ) > > > > Reason for revert: > > Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6). > > > > Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer. > > > > Original issue's description: > > > Enable GN check for webrtc/base > > > > > > It's not possible to enable it for the rtc_base_approved > > > target but since a larger refactoring is ongoing for webrtc/base > > > this CL doesn't attempt to fix that. > > > > > > Changes made: > > > * Move webrtc/system_wrappers/include/stringize_macros.h into > > > webrtc/base:rtc_base_approved_unittests (and corresponding > > > unit test to rtc_base_approved_unittests). > > > * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target. > > > * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into > > > webrtc/base. > > > * Remove unused use include of webrtc/base/fileutils.h in > > > webrtc/base/pathutils.cc > > > > > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > > NOTRY=True > > > > > > Review-Url: https://codereview.webrtc.org/2717083002 > > > Cr-Commit-Position: refs/heads/master@{#17766} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/ed754e71ae8866db641677073274e86fe704eeac > > > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838683002 > > Cr-Commit-Position: refs/heads/master@{#17849} > > Committed: https://chromium.googlesource.com/external/webrtc/+/11ed366c487a938815cd52ad2ab5467b0f90e1ae > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > Review-Url: https://codereview.webrtc.org/2840453004 > Cr-Commit-Position: refs/heads/master@{#17876} > Committed: https://chromium.googlesource.com/external/webrtc/+/7054085e59c3123613cd0014bedb0fe91a56e26f TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828, webrtc:3806, webrtc:7480 Review-Url: https://codereview.webrtc.org/2846483002 Cr-Commit-Position: refs/heads/master@{#17877}
2017-04-26 00:38:48 -07:00
"../../common_audio",
"../../test:rtp_test_utils",
"../rtp_rtcp",
]
public_deps = [
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_tools_minimal",
Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ ) Reason for revert: It causes a Chromium build error: ERROR at //third_party/webrtc/test/BUILD.gn:113:5: Can't load input file. "//third_party/gflags", Original issue's description: > Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ ) > > Reason for revert: > Try to fix the webrtc/test/fuzzers issue and reland this CL because it > contains lots of fixes for our BUILD.gn files. > > Original issue's description: > > Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ ) > > > > Reason for revert: > > Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6). > > > > Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer. > > > > Original issue's description: > > > Enable GN check for webrtc/base > > > > > > It's not possible to enable it for the rtc_base_approved > > > target but since a larger refactoring is ongoing for webrtc/base > > > this CL doesn't attempt to fix that. > > > > > > Changes made: > > > * Move webrtc/system_wrappers/include/stringize_macros.h into > > > webrtc/base:rtc_base_approved_unittests (and corresponding > > > unit test to rtc_base_approved_unittests). > > > * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target. > > > * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into > > > webrtc/base. > > > * Remove unused use include of webrtc/base/fileutils.h in > > > webrtc/base/pathutils.cc > > > > > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > > NOTRY=True > > > > > > Review-Url: https://codereview.webrtc.org/2717083002 > > > Cr-Commit-Position: refs/heads/master@{#17766} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/ed754e71ae8866db641677073274e86fe704eeac > > > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838683002 > > Cr-Commit-Position: refs/heads/master@{#17849} > > Committed: https://chromium.googlesource.com/external/webrtc/+/11ed366c487a938815cd52ad2ab5467b0f90e1ae > > TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828, webrtc:3806, webrtc:7480 > > Review-Url: https://codereview.webrtc.org/2840453004 > Cr-Commit-Position: refs/heads/master@{#17876} > Committed: https://chromium.googlesource.com/external/webrtc/+/7054085e59c3123613cd0014bedb0fe91a56e26f TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828, webrtc:3806, webrtc:7480 Review-Url: https://codereview.webrtc.org/2846483002 Cr-Commit-Position: refs/heads/master@{#17877}
2017-04-26 00:38:48 -07:00
]
if (rtc_enable_protobuf) {
sources += [
"neteq/tools/neteq_packet_source_input.cc",
"neteq/tools/neteq_packet_source_input.h",
]
deps += [ ":rtc_event_log_source" ]
}
}
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_test_tools_deprecated") {
GN migration of target audio_coding/neteq/neteq_test_tools Review-Url: https://codereview.webrtc.org/2178353002 Cr-Commit-Position: refs/heads/master@{#13531}
2016-07-26 06:36:03 -07:00
testonly = true
sources = [
"neteq/test/NETEQTEST_DummyRTPpacket.cc",
"neteq/test/NETEQTEST_DummyRTPpacket.h",
"neteq/test/NETEQTEST_RTPpacket.cc",
"neteq/test/NETEQTEST_RTPpacket.h",
]
deps = [
":cng",
":g711",
":g722",
":ilbc",
":isac",
":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",
GN migration of target audio_coding/neteq/neteq_test_tools Review-Url: https://codereview.webrtc.org/2178353002 Cr-Commit-Position: refs/heads/master@{#13531}
2016-07-26 06:36:03 -07:00
"../..:webrtc_common",
"//testing/gtest",
]
include_dirs = [
"neteq/include",
"neteq/test",
"../../",
]
if (is_win) {
cflags = [
# Disable warnings to enable Win64 build, issue 1323.
"/wd4267", # size_t to int truncation
]
}
}
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 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
config("RTPencode_config") {
defines = [
"CODEC_ILBC",
"CODEC_PCM16B",
"CODEC_G711",
"CODEC_G722",
"CODEC_ISAC",
"CODEC_PCM16B_WB",
"CODEC_ISAC_SWB",
"CODEC_PCM16B_32KHZ",
"CODEC_PCM16B_48KHZ",
"CODEC_CNGCODEC8",
"CODEC_CNGCODEC16",
"CODEC_CNGCODEC32",
"CODEC_ATEVENT_DECODE",
"CODEC_RED",
"CODEC_OPUS",
]
}
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("RTPencode") {
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
deps = [
# TODO(hlundin): Make RTPencode use ACM to encode files.
":cng",
":g711",
":g722",
":ilbc",
":isac",
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
":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_deprecated",
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
":pcm16b",
":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",
"../../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
"../../common_audio",
]
configs += [ ":RTPencode_config" ]
sources = [
"neteq/test/RTPencode.cc",
]
include_dirs = [
"neteq/include",
"neteq/test",
]
if (is_win) {
cflags = [
# Disable warnings to enable Win64 build, issue 1323.
"/wd4267", # size_t to int truncation
]
}
}
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("RTPchange") {
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
sources = [
"neteq/test/RTPchange.cc",
]
deps = [
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_deprecated",
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 = [
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
"../../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("RTPtimeshift") {
Migrated GN target RTPtimeshift. NOTRY=true BUG=6162 Review-Url: https://codereview.webrtc.org/2201883004 Cr-Commit-Position: refs/heads/master@{#13607}
2016-08-02 06:06:33 -07:00
testonly = true
sources = [
"neteq/test/RTPtimeshift.cc",
]
deps = [
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_deprecated",
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",
Migrated GN target RTPtimeshift. NOTRY=true BUG=6162 Review-Url: https://codereview.webrtc.org/2201883004 Cr-Commit-Position: refs/heads/master@{#13607}
2016-08-02 06:06:33 -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("RTPjitter") {
Gyp->GN migration of RTPjitter. BUG=6162 NOTRY=True Review-Url: https://codereview.webrtc.org/2196283002 Cr-Commit-Position: refs/heads/master@{#13620}
2016-08-03 01:17:25 -07:00
testonly = true
deps = [
"../..:webrtc_common",
This is a resubmission of https://codereview.webrtc.org/2047513002/ Original description: Add proper lifetime of encoder-specific settings. Permits passing VideoEncoderConfig between threads and not worry about the lifetime of an underlying void pointer. Also adds type safety to unpacking of codec-specific settings. These settings are not yet propagating to VideoEncoder interfaces, but the aim is to get rid of webrtc::VideoCodec for VideoEncoder. BUG=webrtc:3424 R=perkj@webrtc.org, pbos@webrtc.org TBR=mflodman@webrtc.org Review-Url: https://codereview.webrtc.org/2347843002 Cr-Commit-Position: refs/heads/master@{#14396}
2016-09-27 03:52:02 -07:00
"../../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",
Gyp->GN migration of RTPjitter. BUG=6162 NOTRY=True Review-Url: https://codereview.webrtc.org/2196283002 Cr-Commit-Position: refs/heads/master@{#13620}
2016-08-03 01:17:25 -07:00
"//testing/gtest",
]
sources = [
"neteq/test/RTPjitter.cc",
]
}
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",
"../../system_wrappers:system_wrappers_default",
"//testing/gtest",
"//third_party/gflags:gflags",
]
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 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",
"//third_party/gflags",
]
}
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",
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",
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",
"../../test:test_support",
"//third_party/gflags",
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",
"../../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",
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",
"//third_party/gflags",
]
}
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",
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
"../../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",
"//third_party/gflags",
]
}
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",
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
"../../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_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",
"//third_party/gflags",
]
}
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
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",
"../../test:test_support",
]
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",
]
}
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",
Moved format_macros.h from rtc_base to rtc_base_approved. BUG=webrtc:3806 NOTRY=True Review-Url: https://codereview.webrtc.org/2272003002 Cr-Commit-Position: refs/heads/master@{#13921}
2016-08-25 04:43:45 -07:00
"../../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",
"../..: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",
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
"../../base:rtc_base_approved",
]
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",
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",
Moved format_macros.h from rtc_base to rtc_base_approved. BUG=webrtc:3806 NOTRY=True Review-Url: https://codereview.webrtc.org/2272003002 Cr-Commit-Position: refs/heads/master@{#13921}
2016-08-25 04:43:45 -07:00
"../../base:rtc_base_approved",
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
"../../common_audio",
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
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
# Skip restricting visibility on mobile platforms since the tests on those
# gets additional generated targets which would require many lines here to
# cover (which would be confusing to read and hard to maintain).
if (!is_android && !is_ios) {
visibility = [ "//webrtc/modules:modules_unittests" ]
}
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",
"audio_network_adaptor/mock/mock_controller.h",
"audio_network_adaptor/mock/mock_controller_manager.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_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/mock/mock_audio_encoder.cc",
"codecs/opus/audio_encoder_opus_unittest.cc",
"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",
"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/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_encoder_interface",
":audio_format_conversion",
":audio_network_adaptor",
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
":builtin_audio_encoder_factory",
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
":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",
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",
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",
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
"../../base:protobuf_utils",
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
"../../base:rtc_base",
"../../base:rtc_base_approved",
"../../base:rtc_base_tests_utils",
"../../common_audio",
"../../system_wrappers:system_wrappers",
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",
"../../test:rtp_test_utils",
"../../test:test_common",
"../../test:test_support",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
defines = audio_coding_defines
if (rtc_enable_protobuf) {
defines += [
"WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP",
"WEBRTC_NETEQ_UNITTEST_BITEXACT",
]
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") {
sources = [
"codecs/audio_decoder.h",
]
deps = [
"../../api/audio_codecs:audio_codecs_api",
]
}
# For backwards compatibility only! Use
# webrtc/api/audio_codecs:builtin_audio_decoder_factory instead.
# TODO(kwiberg): Remove this.
rtc_source_set("builtin_audio_decoder_factory") {
sources = [
"codecs/builtin_audio_decoder_factory.h",
]
deps = [
"../../api/audio_codecs:builtin_audio_decoder_factory",
]
}
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