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webrtc_m130/modules/audio_coding/BUILD.gn

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GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../../webrtc.gni")
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
import("audio_coding.gni")
Import proto_library.gni when rtc_enable_protobuf is true Import proto_library.gni when rtc_enable_protobuf is true instead of when build_with_mozilla is false. Makes it maybe easier to reason about the intention (e.g. intention is to not compile any protobuf in, hence flag rtc_enable_protobuf) The build file could not work if build_with_mozilla = true but rtc_enable_protobuf = true. Bug: webrtc:10338 Change-Id: I26e5983bd1519aa46c308b11796d518de5ef7597 Reviewed-on: https://webrtc-review.googlesource.com/c/123763 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26868}
2019-02-26 11:46:17 +02:00
if (rtc_enable_protobuf) {
Update gn files to support Mozilla build Bug: webrtc:8670 No-Presubmit: true Change-Id: I085dc63daa8274b5068540cbf56b6330f40643fa Reviewed-on: https://webrtc-review.googlesource.com/38920 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21624}
2018-01-15 10:20:00 -05:00
import("//third_party/protobuf/proto_library.gni")
}
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility = [ ":*" ]
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") {
Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h Bug: webrtc:5876 Change-Id: I0c92f9410fcf0832bfa321229b3437134255dba6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128085 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27190}
2019-03-19 14:10:16 +01:00
visibility += [ "*" ]
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
sources = [
"include/audio_coding_module_typedefs.h",
]
deps = [
Move NetworkStatistics and AudioDecodingCallStats from common_types.h Bug: webrtc:7626 Change-Id: I1b933b8be7acbca1f1043a374a7cafb95aa9ffde Reviewed-on: https://webrtc-review.googlesource.com/c/111249 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25688}
2018-11-19 11:09:14 +01:00
"../../rtc_base:deprecation",
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
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_coding") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
audio_coding: remove "main" directory This is the last piece of the old directory layout of the modules. Duplicated header files are left in audio_coding/main/include until downstream code is updated to the new location. They have pragma warnings added to them and identical header guards as the new headers to avoid breaking things. BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1481493004 Cr-Commit-Position: refs/heads/master@{#10803}
2015-11-26 04:44:54 -08:00
"acm2/acm_receiver.cc",
"acm2/acm_receiver.h",
"acm2/acm_resampler.cc",
"acm2/acm_resampler.h",
"acm2/audio_coding_module.cc",
"acm2/call_statistics.cc",
"acm2/call_statistics.h",
"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 = []
AudioCodingModule: Remove support for creating encoders After this CL, all audio encoders have to be injected by the caller. This means that there is no special "built-in" set of codecs, and users won't have to pay the binary size and security costs of codecs they aren't using. Bug: webrtc:8396 Change-Id: Idb0959ce395940c8bb3bbb49256cdcd84fc87bb6 Reviewed-on: https://webrtc-review.googlesource.com/c/103821 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25600}
2018-11-12 14:21:58 +01:00
deps = [
":audio_coding_module_typedefs",
":neteq",
"..:module_api",
"..:module_api_public",
"../../api:array_view",
Move rtc::FunctionView to the public API Bug: webrtc:10138 Change-Id: Icc25a2a277a9608701aaddd546882366739991ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27227}
2019-03-21 14:37:36 +01:00
"../../api:function_view",
AudioCodingModule: Remove support for creating encoders After this CL, all audio encoders have to be injected by the caller. This means that there is no special "built-in" set of codecs, and users won't have to pay the binary size and security costs of codecs they aren't using. Bug: webrtc:8396 Change-Id: Idb0959ce395940c8bb3bbb49256cdcd84fc87bb6 Reviewed-on: https://webrtc-review.googlesource.com/c/103821 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25600}
2018-11-12 14:21:58 +01:00
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
Move NetEq headers to api/ This CL also introduces NetEqFactory and NetEqControllerFactory interfaces, as well as several convenience classes for working with them: DefaultNetEqFactory, DefaultNetEqControllerFactory and CustomNetEqFactory. Bug: webrtc:11005 Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29671}
2019-10-31 14:38:11 +01:00
"../../api/neteq:custom_neteq_factory",
"../../api/neteq:default_neteq_controller_factory",
"../../api/neteq:neteq_api",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_audio",
AudioCodingModule: Remove support for creating encoders After this CL, all audio encoders have to be injected by the caller. This means that there is no special "built-in" set of codecs, and users won't have to pay the binary size and security costs of codecs they aren't using. Bug: webrtc:8396 Change-Id: Idb0959ce395940c8bb3bbb49256cdcd84fc87bb6 Reviewed-on: https://webrtc-review.googlesource.com/c/103821 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25600}
2018-11-12 14:21:58 +01:00
"../../common_audio:common_audio_c",
"../../rtc_base:audio_format_to_string",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("legacy_encoded_audio_frame") {
Move the AudioDecoder interface out of NetEq It belongs with the codecs, next to the AudioEncoder interface. R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798 and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799 Review URL: https://webrtc-codereview.appspot.com/27309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:12:53 +00:00
sources = [
Moved legacy_encoded_audio_frame into audio_decoder_interface. audio_decoder.cc depends on LegacyEncodedAudioFrame and LegacyEncodedAudioFrame depends on AudioDecoder::EncodedAudioFrame, so there's no clear way to separate them as of now. This error is also hodling up builds downstream. I expect we'll revisit these dependencies as part of the upcoming larger restructuring effort. NOTRY=true BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2359763002 Cr-Commit-Position: refs/heads/master@{#14329}
2016-09-21 05:50:37 -07:00
"codecs/legacy_encoded_audio_frame.cc",
"codecs/legacy_encoded_audio_frame.h",
Move the AudioDecoder interface out of NetEq It belongs with the codecs, next to the AudioEncoder interface. R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798 and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799 Review URL: https://webrtc-codereview.appspot.com/27309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:12:53 +00:00
]
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("webrtc_cng") {
Hide the AudioEncoderCng class behind a create function And put codecs/cng/webrtc_cng.h in a non-public build target while we're at it. Bug: webrtc:8396 Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c Reviewed-on: https://webrtc-review.googlesource.com/c/108782 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25486}
2018-11-01 11:13:44 +01:00
visibility += webrtc_default_visibility
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
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
]
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 = [
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Hide the AudioEncoderCng class behind a create function And put codecs/cng/webrtc_cng.h in a non-public build target while we're at it. Bug: webrtc:8396 Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c Reviewed-on: https://webrtc-review.googlesource.com/c/108782 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25486}
2018-11-01 11:13:44 +01:00
"../../rtc_base:safe_conversions",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_encoder_cng") {
Hide the AudioEncoderCng class behind a create function And put codecs/cng/webrtc_cng.h in a non-public build target while we're at it. Bug: webrtc:8396 Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c Reviewed-on: https://webrtc-review.googlesource.com/c/108782 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25486}
2018-11-01 11:13:44 +01:00
visibility += [ "*" ]
sources = [
"codecs/cng/audio_encoder_cng.cc",
"codecs/cng/audio_encoder_cng.h",
]
deps = [
":webrtc_cng",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../rtc_base:checks",
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
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("red") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
sources = [
"codecs/red/audio_encoder_copy_red.cc",
"codecs/red/audio_encoder_copy_red.h",
]
deps = [
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"../../api:array_view",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../../api/audio_codecs:audio_codecs_api",
Make the audio codecs' GN targets self-sufficient Also running "gn format" on the file. R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1494993002 . Cr-Commit-Position: refs/heads/master@{#10886}
2015-12-03 17:47:21 +01:00
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("g711") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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
]
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
":legacy_encoded_audio_frame",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"../../api:array_view",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
public_deps = [
":g711_c",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("g711_c") {
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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/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 = [
Move g711 to proper third_party directory Bug: webrtc:8366 Change-Id: Ic57bd5c5c01871aee2956b2a098a79b106f54c9e Reviewed-on: https://webrtc-review.googlesource.com/85375 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24095}
2018-07-25 12:10:22 +02:00
"../third_party/g711:g711_3p",
Extract third party part of g711 codec into separate target Bug: webrtc:8366 Change-Id: I34c7ea707213e0c1a50826896da01f70c072eae5 Reviewed-on: https://webrtc-review.googlesource.com/84741 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23724}
2018-06-21 16:34:10 +02:00
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("g722") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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
]
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
":legacy_encoded_audio_frame",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"../../api:array_view",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
G722 implementation of the AudioEncoderFactoryTemplate API Now the templated AudioEncoderFactory can create G722 encoders! BUG=webrtc:7833 Review-Url: https://codereview.webrtc.org/2934833002 Cr-Commit-Position: refs/heads/master@{#18644}
2017-06-17 17:41:59 -07:00
"../../api/audio_codecs/g722:audio_encoder_g722_config",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
public_deps = [
":g722_c",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("g722_c") {
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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/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 = [
Move g722 to proper third_party directory Bug: webrtc:8366 Change-Id: I81b051dd25da2d7eaa2902af284d8b669ad8e3c9 Reviewed-on: https://webrtc-review.googlesource.com/85620 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24096}
2018-07-25 13:13:44 +02:00
"../third_party/g722:g722_3p",
Extract third party part of g722 codec into separate target Bug: webrtc:8366 Change-Id: I7e08aa53424afd3001f4c22be270a8b0ff7af565 Reviewed-on: https://webrtc-review.googlesource.com/84744 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23725}
2018-06-21 16:39:37 +02:00
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("ilbc") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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
]
deps = [
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
":legacy_encoded_audio_frame",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"../../api:array_view",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Expose ILBC codec in webrtc/api/audio_codecs/ BUG=webrtc:7834, webrtc:7840 Review-Url: https://codereview.webrtc.org/2951873002 Cr-Commit-Position: refs/heads/master@{#18803}
2017-06-28 02:05:04 -07:00
"../../api/audio_codecs/ilbc:audio_encoder_ilbc_config",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
]
public_deps = [
":ilbc_c",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("ilbc_c") {
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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/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",
]
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 = [
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Make the audio codecs' GN targets self-sufficient Also running "gn format" on the file. R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1494993002 . Cr-Commit-Position: refs/heads/master@{#10886}
2015-12-03 17:47:21 +01:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:sanitizer",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
"../../rtc_base/system:unused",
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
}
Delete support for enabling adaptive isac mode This appears unused. If deleted, other code related to isac bandwidth estimation becomes unused and may be deleted in followup cls. Bug: webrtc:10098 Change-Id: Ifeac2e90de895b12c337ea28cc33704350b9abf4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153667 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29252}
2019-09-19 15:21:54 +02:00
rtc_source_set("isac_common") {
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
iSAC: Make separate AudioEncoder and AudioDecoder objects The only shared state is now the bandwidth estimation info. This reduces the amount and complexity of the locking substantially. Review URL: https://codereview.webrtc.org/1208993010 Cr-Commit-Position: refs/heads/master@{#9762}
2015-08-24 02:03:23 -07:00
sources = [
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"codecs/isac/audio_decoder_isac_t.h",
"codecs/isac/audio_decoder_isac_t_impl.h",
iSAC: Make separate AudioEncoder and AudioDecoder objects The only shared state is now the bandwidth estimation info. This reduces the amount and complexity of the locking substantially. Review URL: https://codereview.webrtc.org/1208993010 Cr-Commit-Position: refs/heads/master@{#9762}
2015-08-24 02:03:23 -07:00
"codecs/isac/audio_encoder_isac_t.h",
"codecs/isac/audio_encoder_isac_t_impl.h",
]
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 = [
Break out the part of the iSAC codec that's used for Voice Activity Detection The audio processing code is using parts of the iSAC codec to do voice activity detection (VAD), but it's undesirable for it to pull in the entire iSAC codec as a dependency. So this CL factors out the parts of iSAC that's needed for VAD to a separate build target. Bug: webrtc:8396 Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e Reviewed-on: https://webrtc-review.googlesource.com/69640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23110}
2018-04-27 04:31:53 +02:00
":isac_bwinfo",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
Injectable audio encoders: BuiltinAudioEncoderFactory This CL contains all the changes made to audio_coding while making audio encoders injectable. Apart from some small changes to webrtcvoiceengine, nothing here is hooked up to the outside world. Those changes will be added to a follow-up CL. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2695243005 Cr-Commit-Position: refs/heads/master@{#17569}
2017-04-06 10:03:21 -07:00
"../../api/audio_codecs:audio_codecs_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Replace rtc::Optional with absl::optional in modules/audio_coding This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_coding' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8 Reviewed-on: https://webrtc-review.googlesource.com/84130 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 13:26:36 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
]
iSAC: Make separate AudioEncoder and AudioDecoder objects The only shared state is now the bandwidth estimation info. This reduces the amount and complexity of the locking substantially. Review URL: https://codereview.webrtc.org/1208993010 Cr-Commit-Position: refs/heads/master@{#9762}
2015-08-24 02:03:23 -07:00
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("isac") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
sources = [
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
"codecs/isac/main/include/audio_decoder_isac.h",
"codecs/isac/main/include/audio_encoder_isac.h",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
"codecs/isac/main/source/audio_decoder_isac.cc",
"codecs/isac/main/source/audio_encoder_isac.cc",
]
deps = [
":isac_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
]
public_deps = [
":isac_c",
]
}
Break out the part of the iSAC codec that's used for Voice Activity Detection The audio processing code is using parts of the iSAC codec to do voice activity detection (VAD), but it's undesirable for it to pull in the entire iSAC codec as a dependency. So this CL factors out the parts of iSAC that's needed for VAD to a separate build target. Bug: webrtc:8396 Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e Reviewed-on: https://webrtc-review.googlesource.com/69640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23110}
2018-04-27 04:31:53 +02:00
rtc_source_set("isac_bwinfo") {
sources = [
"codecs/isac/bandwidth_info.h",
]
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
deps = []
Break out the part of the iSAC codec that's used for Voice Activity Detection The audio processing code is using parts of the iSAC codec to do voice activity detection (VAD), but it's undesirable for it to pull in the entire iSAC codec as a dependency. So this CL factors out the parts of iSAC that's needed for VAD to a separate build target. Bug: webrtc:8396 Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e Reviewed-on: https://webrtc-review.googlesource.com/69640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23110}
2018-04-27 04:31:53 +02:00
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("isac_vad") {
Break out the part of the iSAC codec that's used for Voice Activity Detection The audio processing code is using parts of the iSAC codec to do voice activity detection (VAD), but it's undesirable for it to pull in the entire iSAC codec as a dependency. So this CL factors out the parts of iSAC that's needed for VAD to a separate build target. Bug: webrtc:8396 Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e Reviewed-on: https://webrtc-review.googlesource.com/69640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23110}
2018-04-27 04:31:53 +02:00
visibility += webrtc_default_visibility
sources = [
"codecs/isac/main/source/filter_functions.c",
"codecs/isac/main/source/filter_functions.h",
"codecs/isac/main/source/isac_vad.c",
"codecs/isac/main/source/isac_vad.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_filter.h",
"codecs/isac/main/source/settings.h",
"codecs/isac/main/source/structs.h",
]
deps = [
":isac_bwinfo",
"../../rtc_base:compile_assert_c",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
Break out the part of the iSAC codec that's used for Voice Activity Detection The audio processing code is using parts of the iSAC codec to do voice activity detection (VAD), but it's undesirable for it to pull in the entire iSAC codec as a dependency. So this CL factors out the parts of iSAC that's needed for VAD to a separate build target. Bug: webrtc:8396 Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e Reviewed-on: https://webrtc-review.googlesource.com/69640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23110}
2018-04-27 04:31:53 +02:00
"../../rtc_base/system:ignore_warnings",
Untangle fft third party lib from dependon WebRTC TBR=phoglund Bug: webrtc:9558 Change-Id: I6cc1936549f008694c3617c1d990524c34da16e3 Reviewed-on: https://webrtc-review.googlesource.com/90411 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24115}
2018-07-26 15:07:33 +02:00
"../third_party/fft",
Break out the part of the iSAC codec that's used for Voice Activity Detection The audio processing code is using parts of the iSAC codec to do voice activity detection (VAD), but it's undesirable for it to pull in the entire iSAC codec as a dependency. So this CL factors out the parts of iSAC that's needed for VAD to a separate build target. Bug: webrtc:8396 Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e Reviewed-on: https://webrtc-review.googlesource.com/69640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23110}
2018-04-27 04:31:53 +02:00
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("isac_c") {
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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",
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/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/spectrum_ar_model_tables.c",
"codecs/isac/main/source/spectrum_ar_model_tables.h",
"codecs/isac/main/source/transform.c",
]
if (is_linux) {
libs = [ "m" ]
}
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 = [
Break out the part of the iSAC codec that's used for Voice Activity Detection The audio processing code is using parts of the iSAC codec to do voice activity detection (VAD), but it's undesirable for it to pull in the entire iSAC codec as a dependency. So this CL factors out the parts of iSAC that's needed for VAD to a separate build target. Bug: webrtc:8396 Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e Reviewed-on: https://webrtc-review.googlesource.com/69640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23110}
2018-04-27 04:31:53 +02:00
":isac_bwinfo",
":isac_vad",
Make an AudioEncoder subclass for iSAC BUG=3926 Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675 and reverted: https://code.google.com/p/webrtc/source/detail?r=7676 R=henrik.lundin@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:08:19 +00:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:compile_assert_c",
"../../rtc_base:rtc_base_approved",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
Untangle fft third party lib from dependon WebRTC TBR=phoglund Bug: webrtc:9558 Change-Id: I6cc1936549f008694c3617c1d990524c34da16e3 Reviewed-on: https://webrtc-review.googlesource.com/90411 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24115}
2018-07-26 15:07:33 +02:00
"../third_party/fft",
Extract fft to separate target to be able to move it to third_party fft.c is third party library and have to be moved to proper third_party directory. So this CL will extract it to separate gn target to be able then to move it to proper location. Bug: webrtc:8366 Change-Id: I228ebab3c821aa7095f7aa460c23c2ea0fb98f01 Reviewed-on: https://webrtc-review.googlesource.com/85640 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23753}
2018-06-26 14:50:13 +02:00
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("isac_fix") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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",
]
deps = [
":isac_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
"../../common_audio",
"../../system_wrappers",
]
public_deps = [
":isac_fix_c",
]
if (rtc_build_with_neon) {
deps += [ ":isac_neon" ]
}
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("isac_fix_common") {
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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
sources = [
"codecs/isac/fix/source/codec.h",
Using fully qualified #include paths in isac code. WebRTC internal code should always used include paths that starts from the root of the project and that clearly identify the header file. This allows 'gn check' to actually keep dependencies under control because 'gn check' cannot enforce anything if the include path is not fully qualified (starting from the root of the project). Bug: webrtc:8815 Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc Reviewed-on: https://webrtc-review.googlesource.com/46101 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21847}
2018-02-01 14:12:55 +01:00
"codecs/isac/fix/source/entropy_coding.h",
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
"codecs/isac/fix/source/fft.c",
"codecs/isac/fix/source/fft.h",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"codecs/isac/fix/source/filterbank_internal.h",
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
"codecs/isac/fix/source/settings.h",
Using fully qualified #include paths in isac code. WebRTC internal code should always used include paths that starts from the root of the project and that clearly identify the header file. This allows 'gn check' to actually keep dependencies under control because 'gn check' cannot enforce anything if the include path is not fully qualified (starting from the root of the project). Bug: webrtc:8815 Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc Reviewed-on: https://webrtc-review.googlesource.com/46101 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21847}
2018-02-01 14:12:55 +01:00
"codecs/isac/fix/source/structs.h",
Moving transform_tables.c to isac_fix_common. The target modules/audio_coding:isac_neon needs to link with transform_tables.c but adding a dependency between isac_neon and isac_fix_c creates a circular dependency. This CL moves transform_tables.c to isac_fix_common (which is already a dependency of isac_neon). Bug: None Change-Id: I4135ec772b0017e77f1411e9a8093b495220c636 Reviewed-on: https://webrtc-review.googlesource.com/71581 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22968}
2018-04-20 13:54:53 -07:00
"codecs/isac/fix/source/transform_tables.c",
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
]
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
deps = [
Break out the part of the iSAC codec that's used for Voice Activity Detection The audio processing code is using parts of the iSAC codec to do voice activity detection (VAD), but it's undesirable for it to pull in the entire iSAC codec as a dependency. So this CL factors out the parts of iSAC that's needed for VAD to a separate build target. Bug: webrtc:8396 Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e Reviewed-on: https://webrtc-review.googlesource.com/69640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23110}
2018-04-27 04:31:53 +02:00
":isac_bwinfo",
Using fully qualified #include paths in isac code. WebRTC internal code should always used include paths that starts from the root of the project and that clearly identify the header file. This allows 'gn check' to actually keep dependencies under control because 'gn check' cannot enforce anything if the include path is not fully qualified (starting from the root of the project). Bug: webrtc:8815 Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc Reviewed-on: https://webrtc-review.googlesource.com/46101 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21847}
2018-02-01 14:12:55 +01:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
]
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
}
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
Moving asm code out of isac_fix_c sources list BUG=webrtc:7743 Review-Url: https://codereview.webrtc.org/2973613002 Cr-Commit-Position: refs/heads/master@{#18911}
2017-07-06 03:48:55 -07:00
rtc_source_set("isac_fix_c_arm_asm") {
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
Moving asm code out of isac_fix_c sources list BUG=webrtc:7743 Review-Url: https://codereview.webrtc.org/2973613002 Cr-Commit-Position: refs/heads/master@{#18911}
2017-07-06 03:48:55 -07:00
sources = []
if (current_cpu == "arm" && arm_version >= 7) {
sources += [
"codecs/isac/fix/source/lattice_armv7.S",
"codecs/isac/fix/source/pitch_filter_armv6.S",
]
Adding :isac_fix_c_arm_asm missing dependency. TBR=phoglund@webrtc.org Bug: None Change-Id: I6cb1a442274a627e03a58098d74c8bbf00e492a3 Reviewed-on: https://webrtc-review.googlesource.com/46100 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21806}
2018-01-30 13:36:55 +01:00
deps = [
Adding missing ASM dependencies. Bug: webrtc:8603 Change-Id: I7b417759fcdd01879029afcc5afc50300016fd72 Reviewed-on: https://webrtc-review.googlesource.com/56840 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22159}
2018-02-22 15:10:27 +01:00
":isac_fix_common",
Move asm_defines.h to rtc_base/system/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. Bug: webrtc:8445 NOPRESUBMIT=true Change-Id: I30d01fcb9cbe1427a7703a3cdd7befae751066b5 Reviewed-on: https://webrtc-review.googlesource.com/21982 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22550}
2018-03-22 02:59:49 +01:00
"../../rtc_base/system:asm_defines",
Adding :isac_fix_c_arm_asm missing dependency. TBR=phoglund@webrtc.org Bug: None Change-Id: I6cb1a442274a627e03a58098d74c8bbf00e492a3 Reviewed-on: https://webrtc-review.googlesource.com/46100 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21806}
2018-01-30 13:36:55 +01:00
]
Moving asm code out of isac_fix_c sources list BUG=webrtc:7743 Review-Url: https://codereview.webrtc.org/2973613002 Cr-Commit-Position: refs/heads/master@{#18911}
2017-07-06 03:48:55 -07:00
}
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("isac_fix_c") {
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
audio_coding: rename interface -> include BUG=webrtc:5095 R=henrik.lundin@webrtc.org TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417173004 . Cr-Commit-Position: refs/heads/master@{#10444}
2015-10-29 11:31:02 +01:00
"codecs/isac/fix/include/audio_decoder_isacfix.h",
"codecs/isac/fix/include/audio_encoder_isacfix.h",
"codecs/isac/fix/include/isacfix.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/arith_routines.c",
"codecs/isac/fix/source/arith_routines_hist.c",
"codecs/isac/fix/source/arith_routines_logist.c",
"codecs/isac/fix/source/arith_routins.h",
"codecs/isac/fix/source/bandwidth_estimator.c",
"codecs/isac/fix/source/bandwidth_estimator.h",
"codecs/isac/fix/source/decode.c",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/isac/fix/source/decode_bwe.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/decode_plc.c",
"codecs/isac/fix/source/encode.c",
"codecs/isac/fix/source/entropy_coding.c",
"codecs/isac/fix/source/filterbank_tables.c",
"codecs/isac/fix/source/filterbank_tables.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/isac/fix/source/filterbanks.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/filters.c",
"codecs/isac/fix/source/initialize.c",
Move AudioDecoderIsac* to its own files Currently, it's sitting in AudioEncoderIsac*'s files, which is less than obvious. This CL puts the encoder and decoder in separate files together with the C implementation; CLs are afoot to make it so for the other built-in codecs as well. BUG=webrtc:4557 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1339253003 . Cr-Commit-Position: refs/heads/master@{#10018}
2015-09-22 19:31:40 +02:00
"codecs/isac/fix/source/isac_fix_type.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/isacfix.c",
"codecs/isac/fix/source/lattice.c",
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
"codecs/isac/fix/source/lattice_c.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/lpc_masking_model.c",
"codecs/isac/fix/source/lpc_masking_model.h",
"codecs/isac/fix/source/lpc_tables.c",
"codecs/isac/fix/source/lpc_tables.h",
"codecs/isac/fix/source/pitch_estimator.c",
"codecs/isac/fix/source/pitch_estimator.h",
Reduce the number of registers used in MIPS optimizations. This change is needed by ChromeOS as it introduces -fno-omit-frame-pointer flag (see code.google.com/p/chromium/issues/detail?id=477749). This causes compile error for MIPS, as some MIPS optimization blocks use maximum possible number of available registers. Also, this change contains minor GN build fix for MIPS platform regarding the pitch_filter_mips.c / pitch_filter_c.c file inclusion. BUG=477749 R=andrew@webrtc.org, djordje.pesut@imgtec.com, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48139004 Patch from Ljubomir Papuga <lpapuga@mips.com>. Cr-Commit-Position: refs/heads/master@{#9047}
2015-04-21 16:52:45 -07:00
"codecs/isac/fix/source/pitch_estimator_c.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/pitch_filter.c",
Reduce the number of registers used in MIPS optimizations. This change is needed by ChromeOS as it introduces -fno-omit-frame-pointer flag (see code.google.com/p/chromium/issues/detail?id=477749). This causes compile error for MIPS, as some MIPS optimization blocks use maximum possible number of available registers. Also, this change contains minor GN build fix for MIPS platform regarding the pitch_filter_mips.c / pitch_filter_c.c file inclusion. BUG=477749 R=andrew@webrtc.org, djordje.pesut@imgtec.com, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48139004 Patch from Ljubomir Papuga <lpapuga@mips.com>. Cr-Commit-Position: refs/heads/master@{#9047}
2015-04-21 16:52:45 -07:00
"codecs/isac/fix/source/pitch_filter_c.c",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"codecs/isac/fix/source/pitch_gain_tables.c",
"codecs/isac/fix/source/pitch_gain_tables.h",
"codecs/isac/fix/source/pitch_lag_tables.c",
"codecs/isac/fix/source/pitch_lag_tables.h",
"codecs/isac/fix/source/spectrum_ar_model_tables.c",
"codecs/isac/fix/source/spectrum_ar_model_tables.h",
"codecs/isac/fix/source/transform.c",
]
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 = [
Break out the part of the iSAC codec that's used for Voice Activity Detection The audio processing code is using parts of the iSAC codec to do voice activity detection (VAD), but it's undesirable for it to pull in the entire iSAC codec as a dependency. So this CL factors out the parts of iSAC that's needed for VAD to a separate build target. Bug: webrtc:8396 Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e Reviewed-on: https://webrtc-review.googlesource.com/69640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23110}
2018-04-27 04:31:53 +02:00
":isac_bwinfo",
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",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:compile_assert_c",
"../../rtc_base:rtc_base_approved",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:sanitizer",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:cpu_features_api",
Untangle fft third party lib from dependon WebRTC TBR=phoglund Bug: webrtc:9558 Change-Id: I6cc1936549f008694c3617c1d990524c34da16e3 Reviewed-on: https://webrtc-review.googlesource.com/90411 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24115}
2018-07-26 15:07:33 +02:00
"../third_party/fft",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
]
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
public_deps = [
":isac_fix_common",
]
Fixing cross-compiling issues on android arm BUG=webrtc:7042 Review-Url: https://codereview.webrtc.org/2647293006 Cr-Commit-Position: refs/heads/master@{#16265}
2017-01-25 05:34:46 -08:00
if (rtc_build_with_neon) {
deps += [ ":isac_neon" ]
Explicitly add -mfpu=neon to all targets that use NEON Remove obsolete comment about Chromium not defining NEON for Android. Semi-related fix: don't use `rtc_remove_configs` directly, `suppressed_configs` is the "public interface". Bug: webrtc:9579 Change-Id: I512628feb462a29432f1356cfef00efe1ddaf84f Reviewed-on: https://webrtc-review.googlesource.com/91761 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24165}
2018-08-01 12:18:05 +02:00
# TODO(bugs.webrtc.org/9579): Consider moving the usage of NEON from
# pitch_estimator_c.c into the "isac_neon" target and delete this flag:
if (current_cpu != "arm64") {
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
}
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
}
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
if (current_cpu == "arm" && arm_version >= 7) {
sources -= [
"codecs/isac/fix/source/lattice_c.c",
"codecs/isac/fix/source/pitch_filter_c.c",
]
Moving asm code out of isac_fix_c sources list BUG=webrtc:7743 Review-Url: https://codereview.webrtc.org/2973613002 Cr-Commit-Position: refs/heads/master@{#18911}
2017-07-06 03:48:55 -07:00
deps += [ ":isac_fix_c_arm_asm" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530) Includes GN changes from https://webrtc-codereview.appspot.com/39249004/ Android changes for JNI were required due to https://codereview.chromium.org/843103003 Other relevant changes: * src/buildtools: 5c5e924..93b3d0a * src/third_party/boringssl/src: d306f16..b180ee9 * src/third_party/icu: 4e3266f..2081ee6 * src/third_party/libvpx: 5cdd302..33bbffe * src/third_party/usrsctp/usrsctplib: 190c8cb..13718c7 * src/tools/gyp: 4d7c139..3464008 * src/tools/swarming_client: bdad118..1b7bfec Details: https://chromium.googlesource.com/chromium/src/+/b0c3ed3..2c3ffb2/DEPS Clang version was not updated in this roll. R=dpranke@chromium.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40079004 Cr-Commit-Position: refs/heads/master@{#8466} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8466 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 19:08:31 +00:00
if (current_cpu == "mipsel") {
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources += [
"codecs/isac/fix/source/entropy_coding_mips.c",
"codecs/isac/fix/source/filters_mips.c",
"codecs/isac/fix/source/lattice_mips.c",
"codecs/isac/fix/source/pitch_estimator_mips.c",
"codecs/isac/fix/source/transform_mips.c",
]
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
sources -= [
"codecs/isac/fix/source/lattice_c.c",
"codecs/isac/fix/source/pitch_estimator_c.c",
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
if (mips_dsp_rev > 0) {
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
sources += [ "codecs/isac/fix/source/filterbanks_mips.c" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
if (mips_dsp_rev > 1) {
sources += [
"codecs/isac/fix/source/lpc_masking_model_mips.c",
"codecs/isac/fix/source/pitch_filter_mips.c",
]
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
sources -= [ "codecs/isac/fix/source/pitch_filter_c.c" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
}
}
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
if (rtc_build_with_neon) {
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("isac_neon") {
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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") {
Explicitly add -mfpu=neon to all targets that use NEON Remove obsolete comment about Chromium not defining NEON for Android. Semi-related fix: don't use `rtc_remove_configs` directly, `suppressed_configs` is the "public interface". Bug: webrtc:9579 Change-Id: I512628feb462a29432f1356cfef00efe1ddaf84f Reviewed-on: https://webrtc-review.googlesource.com/91761 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24165}
2018-08-01 12:18:05 +02:00
# Enable compilation for the NEON instruction set.
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" ]
}
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
":isac_fix_common",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("pcm16b") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
Move AudioDecoderPcm16B next to AudioEncoderPcm16B All AudioDecoder subclasses have historically lived in NetEq, but they fit better with the codec they wrap. BUG=webrtc:4557 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1348113002 . Cr-Commit-Position: refs/heads/master@{#9963}
2015-09-17 07:47:34 +02:00
"codecs/pcm16b/audio_decoder_pcm16b.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/pcm16b/audio_decoder_pcm16b.h",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"codecs/pcm16b/audio_encoder_pcm16b.cc",
modules/audio_coding: Remove some codec include dirs Also clean up some include_dir entries and update the few references to them with absolute include paths instead. Finally fixed a few lint errors and invalid header guards. None of these are used downstream. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1438663003 . Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 23:07:57 +01:00
"codecs/pcm16b/audio_encoder_pcm16b.h",
L16 implementation of the Audio{En,De}coderFactoryTemplate APIs (This is a re-land of https://codereview.webrtc.org/2995523002, with a narrowing conversion removed.) BUG=webrtc:7836, webrtc:7842 Review-Url: https://codereview.webrtc.org/2995993002 Cr-Commit-Position: refs/heads/master@{#19369}
2017-08-16 06:12:57 -07:00
"codecs/pcm16b/pcm16b_common.cc",
"codecs/pcm16b/pcm16b_common.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
Make an AudioEncoder subclass for PCM16B The implementation depends on AudioEncoderPcm to reduce code duplication. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7872 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:47:19 +00:00
deps = [
":g711",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
":legacy_encoded_audio_frame",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"../../api:array_view",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Make an AudioEncoder subclass for PCM16B The implementation depends on AudioEncoderPcm to reduce code duplication. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7872 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:47:19 +00:00
]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
public_deps = [
":pcm16b_c",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("pcm16b_c") {
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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/pcm16b/pcm16b.c",
"codecs/pcm16b/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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_coding_opus_common") {
Decoder for multistream Opus. See https://webrtc-review.googlesource.com/c/src/+/121764 for the overall vision. This CL adds a multistream Opus decoder. It's a new code-path to not interfere with the standard Opus decoder. We introduce new SDP syntax, which uses terminology of RFC 7845. We also set up the decoder side to parse it. The encoder part will come in a later CL. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 17:19:41 +02:00
sources = [
"codecs/opus/audio_coder_opus_common.cc",
"codecs/opus/audio_coder_opus_common.h",
]
deps = [
"../../api:array_view",
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:checks",
"../../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("webrtc_opus") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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 = [
Decoder for multistream Opus. See https://webrtc-review.googlesource.com/c/src/+/121764 for the overall vision. This CL adds a multistream Opus decoder. It's a new code-path to not interfere with the standard Opus decoder. We introduce new SDP syntax, which uses terminology of RFC 7845. We also set up the decoder side to parse it. The encoder part will come in a later CL. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 17:19:41 +02:00
":audio_coding_opus_common",
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",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"../../api:array_view",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
Opus implementation of the AudioEncoderFactoryTemplate API This was previously reverted, because external projects were using the internal webrtc::AudioEncoderOpus class and broke when it was renamed. This re-land avoids renaming it immediately, to give those projects time to adapt. It also has to revert some of the changes I had made to the Config struct, since that was also used by the same external projects. BUG=webrtc:7831 Review-Url: https://codereview.webrtc.org/2948483002 Cr-Commit-Position: refs/heads/master@{#18852}
2017-06-30 04:23:22 -07:00
"../../api/audio_codecs/opus:audio_encoder_opus_config",
Update smoothed bitrate. BUG=webrtc:6443 Review-Url: https://codereview.webrtc.org/2546493002 Cr-Commit-Position: refs/heads/master@{#16036}
2017-01-12 10:17:38 -08:00
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove webrtc::ProtoString. Bug: None Change-Id: If99a977532eda41eada25f57ff0ff6fe17085986 Reviewed-on: https://webrtc-review.googlesource.com/c/122581 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26726}
2019-02-16 09:59:29 +01:00
"../../rtc_base:protobuf_utils",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../../rtc_base:safe_minmax",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:field_trial",
Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. Bug: webrtc:5876 Change-Id: Ica2d47ca45b8ef01a548d8dbe31dbed740a0ebda Reviewed-on: https://webrtc-review.googlesource.com/c/106820 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25306}
2018-10-22 09:48:08 +02:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional in modules/audio_coding This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_coding' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8 Reviewed-on: https://webrtc-review.googlesource.com/84130 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 13:26:36 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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
]
Decoder for multistream Opus. See https://webrtc-review.googlesource.com/c/src/+/121764 for the overall vision. This CL adds a multistream Opus decoder. It's a new code-path to not interfere with the standard Opus decoder. We introduce new SDP syntax, which uses terminology of RFC 7845. We also set up the decoder side to parse it. The encoder part will come in a later CL. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 17:19:41 +02:00
public_deps = [ # no-presubmit-check TODO(webrtc:8603)
Renaming opus_interface.c to opus_interface.cc. This is to allow advanced features of WebRTC/Chrome e.g., field trials. More style compliant changes may follow up. Only a minimal (not in terms of line changes) is applied, so that presubmit does not complain. These changes include 1. removing unused headers. 2. eliminating c-style casting. Bug: b/143582588 Change-Id: I6d0fd926c542ab0afdc38cc4bf03aaf584ec13dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158670 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29657}
2019-10-29 21:36:13 +01:00
":webrtc_opus_wrapper",
Decoder for multistream Opus. See https://webrtc-review.googlesource.com/c/src/+/121764 for the overall vision. This CL adds a multistream Opus decoder. It's a new code-path to not interfere with the standard Opus decoder. We introduce new SDP syntax, which uses terminology of RFC 7845. We also set up the decoder side to parse it. The encoder part will come in a later CL. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 17:19:41 +02:00
]
defines = audio_codec_defines
if (rtc_build_opus) {
public_deps += [ rtc_opus_dir ] # no-presubmit-check TODO(webrtc:8603)
} else if (build_with_mozilla) {
include_dirs = [ "/media/libopus/include" ]
}
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("webrtc_multiopus") {
Decoder for multistream Opus. See https://webrtc-review.googlesource.com/c/src/+/121764 for the overall vision. This CL adds a multistream Opus decoder. It's a new code-path to not interfere with the standard Opus decoder. We introduce new SDP syntax, which uses terminology of RFC 7845. We also set up the decoder side to parse it. The encoder part will come in a later CL. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 17:19:41 +02:00
visibility += webrtc_default_visibility
poisonous = [ "audio_codecs" ]
sources = [
"codecs/opus/audio_decoder_multi_channel_opus_impl.cc",
"codecs/opus/audio_decoder_multi_channel_opus_impl.h",
Encoder side of Multistream Opus. Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely. Adds an ENCODER and sets it up to parse SDP config for multistream opus. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27775}
2019-04-25 15:09:32 +02:00
"codecs/opus/audio_encoder_multi_channel_opus_impl.cc",
"codecs/opus/audio_encoder_multi_channel_opus_impl.h",
Decoder for multistream Opus. See https://webrtc-review.googlesource.com/c/src/+/121764 for the overall vision. This CL adds a multistream Opus decoder. It's a new code-path to not interfere with the standard Opus decoder. We introduce new SDP syntax, which uses terminology of RFC 7845. We also set up the decoder side to parse it. The encoder part will come in a later CL. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 17:19:41 +02:00
]
deps = [
":audio_coding_opus_common",
"../../api/audio_codecs:audio_codecs_api",
Encoder side of Multistream Opus. Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely. Adds an ENCODER and sets it up to parse SDP config for multistream opus. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27775}
2019-04-25 15:09:32 +02:00
"../../api/audio_codecs/opus:audio_decoder_opus_config",
Decoder for multistream Opus. See https://webrtc-review.googlesource.com/c/src/+/121764 for the overall vision. This CL adds a multistream Opus decoder. It's a new code-path to not interfere with the standard Opus decoder. We introduce new SDP syntax, which uses terminology of RFC 7845. We also set up the decoder side to parse it. The encoder part will come in a later CL. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 17:19:41 +02:00
"../../api/audio_codecs/opus:audio_encoder_opus_config",
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../rtc_base:macromagic",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:safe_minmax",
"../../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/memory",
Encoder side of Multistream Opus. Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely. Adds an ENCODER and sets it up to parse SDP config for multistream opus. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27775}
2019-04-25 15:09:32 +02:00
"//third_party/abseil-cpp/absl/strings",
Decoder for multistream Opus. See https://webrtc-review.googlesource.com/c/src/+/121764 for the overall vision. This CL adds a multistream Opus decoder. It's a new code-path to not interfere with the standard Opus decoder. We introduce new SDP syntax, which uses terminology of RFC 7845. We also set up the decoder side to parse it. The encoder part will come in a later CL. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 17:19:41 +02:00
"//third_party/abseil-cpp/absl/types:optional",
]
public_deps = [ # no-presubmit-check TODO(webrtc:8603)
Renaming opus_interface.c to opus_interface.cc. This is to allow advanced features of WebRTC/Chrome e.g., field trials. More style compliant changes may follow up. Only a minimal (not in terms of line changes) is applied, so that presubmit does not complain. These changes include 1. removing unused headers. 2. eliminating c-style casting. Bug: b/143582588 Change-Id: I6d0fd926c542ab0afdc38cc4bf03aaf584ec13dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158670 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29657}
2019-10-29 21:36:13 +01:00
":webrtc_opus_wrapper",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
]
Make an AudioEncoder subclass for RED This class only supports the simple case of payload duplication. That is, one single encoder is used, and the redundant payload is a one-step delayed payload. BUG=3926 R=kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 13:41:36 +00:00
Add 120ms frame ability to ANA BUG=webrtc:7093 Review-Url: https://codereview.webrtc.org/2669733002 Cr-Commit-Position: refs/heads/master@{#16416}
2017-02-02 07:47:19 -08:00
defines = audio_codec_defines
Let Opus increase complexity for low bitrates This change adds code that lets Opus increase the complexity setting at low bitrates (only relevant for mobile where the default complexity is not already maximum). The feature is default off. Also adding a performance test to make sure the complexity adaptation has desired effect. BUG=webrtc:6708 Review-Url: https://codereview.webrtc.org/2503443002 Cr-Commit-Position: refs/heads/master@{#15182}
2016-11-22 02:07:54 -08:00
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
if (rtc_build_opus) {
public_deps += [ rtc_opus_dir ]
} else if (build_with_mozilla) {
Update gn files to support Mozilla build Bug: webrtc:8670 No-Presubmit: true Change-Id: I085dc63daa8274b5068540cbf56b6330f40643fa Reviewed-on: https://webrtc-review.googlesource.com/38920 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21624}
2018-01-15 10:20:00 -05:00
include_dirs = [ "/media/libopus/include" ]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
}
}
Renaming opus_interface.c to opus_interface.cc. This is to allow advanced features of WebRTC/Chrome e.g., field trials. More style compliant changes may follow up. Only a minimal (not in terms of line changes) is applied, so that presubmit does not complain. These changes include 1. removing unused headers. 2. eliminating c-style casting. Bug: b/143582588 Change-Id: I6d0fd926c542ab0afdc38cc4bf03aaf584ec13dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158670 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29657}
2019-10-29 21:36:13 +01:00
rtc_library("webrtc_opus_wrapper") {
GN hack to tag targets as poisonous (and use it with audio codecs) Only specially taggged targets may transitively depend on poisonous targets. We first apply it to audio codecs. This makes it much clearer exactly what parts of the code still have dependencies on the audio codecs (and we want to eventually get rid of pretty much all of them). Bug: webrtc:8396, webrtc:9121 Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43 Reviewed-on: https://webrtc-review.googlesource.com/69520 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-22 19:55:00 +02:00
poisonous = [ "audio_codecs" ]
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/opus/opus_inst.h",
Renaming opus_interface.c to opus_interface.cc. This is to allow advanced features of WebRTC/Chrome e.g., field trials. More style compliant changes may follow up. Only a minimal (not in terms of line changes) is applied, so that presubmit does not complain. These changes include 1. removing unused headers. 2. eliminating c-style casting. Bug: b/143582588 Change-Id: I6d0fd926c542ab0afdc38cc4bf03aaf584ec13dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158670 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29657}
2019-10-29 21:36:13 +01:00
"codecs/opus/opus_interface.cc",
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/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 = [
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Move rtc_base/ignore_wundef.h to its own target. Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to move it to its own target and allow its dependant to avoid to take a dependency rtc_base:on rtc_base_approved. Bug: webrtc:9419 Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29548}
2019-10-19 12:03:23 +02:00
"../../rtc_base:ignore_wundef",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Make Opus PLC always output 10ms audio. BUG: b/143582588 Change-Id: I41ad5f4f91d9af3f595666a8f32b7ab5382605bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158672 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29733}
2019-11-04 14:47:52 +01:00
"../../system_wrappers:field_trial",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
if (rtc_enable_protobuf) {
proto_library("ana_debug_dump_proto") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
sources = [
"audio_network_adaptor/debug_dump.proto",
]
use link_deps in ana_debug_dump_proto I will deprecate deps in proto_library for improved build throughput. We can use link_deps here instead. Bug: chromium:938011 Change-Id: Iafa83000c3f7f9ffdc0c376a2297b4a9380b7594 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125820 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Takuto Ikuta <tikuta@google.com> Cr-Commit-Position: refs/heads/master@{#26989}
2019-03-06 17:53:55 +09:00
link_deps = [ ":ana_config_proto" ]
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
proto_out_dir = "modules/audio_coding/audio_network_adaptor"
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
}
Creating controller manager from config string in audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2364403004 Cr-Commit-Position: refs/heads/master@{#14466}
2016-10-02 14:53:37 -07:00
proto_library("ana_config_proto") {
Add visibility to ana config proto Downstream projects need to be able to configure ANA without hacking or redefining protos. Bug: webrtc:9719 Change-Id: Idd80471066ff41a9265adbdb738cc98cc97b2e6e Reviewed-on: https://webrtc-review.googlesource.com/c/110765 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25629}
2018-11-13 10:13:42 -08:00
visibility += [ "*" ]
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
sources = [
"audio_network_adaptor/config.proto",
]
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
proto_out_dir = "modules/audio_coding/audio_network_adaptor"
Creating controller manager from config string in audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2364403004 Cr-Commit-Position: refs/heads/master@{#14466}
2016-10-02 14:53:37 -07:00
}
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_network_adaptor_config") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Break the ANA build-target into ANA and ANA-config This is done to solve a dependency-cycle with the RtcEventLog - now the RtcEventLog can depend on the config part of ANA, and be able to peer inside, while the implementation part of ANA can invoke the RtcEventLog. BUG=webrtc:8111 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/3010343002 Cr-Commit-Position: refs/heads/master@{#19793}
2017-09-12 04:38:25 -07:00
sources = [
"audio_network_adaptor/audio_network_adaptor_config.cc",
"audio_network_adaptor/include/audio_network_adaptor_config.h",
]
deps = [
Replace rtc::Optional with absl::optional in modules/audio_coding This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_coding' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8 Reviewed-on: https://webrtc-review.googlesource.com/84130 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 13:26:36 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Break the ANA build-target into ANA and ANA-config This is done to solve a dependency-cycle with the RtcEventLog - now the RtcEventLog can depend on the config part of ANA, and be able to peer inside, while the implementation part of ANA can invoke the RtcEventLog. BUG=webrtc:8111 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/3010343002 Cr-Commit-Position: refs/heads/master@{#19793}
2017-09-12 04:38:25 -07:00
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_network_adaptor") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Adding AudioNetworkAdaptor interfaces. AudioNetworkAdaptor is supposed to facilitate AudioEncoder to adapt to varying network conditions. This is the first of a sequence of CLs that are to add one implementation of AudioNetworkAdaptor. This CL illustrates the interfaces of the AudioNetworkAdaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2308573002 Cr-Commit-Position: refs/heads/master@{#14115}
2016-09-07 13:51:51 -07:00
sources = [
Adding basic implementation of AudioNetworkAdaptor. The basic implementation of AudioNetworkAdaptor include the introduction of 1. Controller 2. ControllerManager ControllerManager is to hold all needed controllers. It also orders them according to their significance in dealing with current network condition. Controller provides an interface MakeDecision, which has to be implemented by specific controllers. AudioNetworkAdaptorImpl calls MakeDecision of the controllers in the order decided by ControllerManager to collect EncoderRuntimeConfig. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2306083002 Cr-Commit-Position: refs/heads/master@{#14201}
2016-09-13 13:34:15 -07:00
"audio_network_adaptor/audio_network_adaptor_impl.cc",
"audio_network_adaptor/audio_network_adaptor_impl.h",
Relanding of "Adding BitrateController to audio network adaptor." Adding BitrateController to audio network adaptor was first landed in https://codereview.webrtc.org/2334613002/ but reverted in https://codereview.webrtc.org/2352223002/ due to ODR violation. This CL tries to use namespace trick to solve the ODR problem. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2353293002 Cr-Commit-Position: refs/heads/master@{#14343}
2016-09-21 23:01:26 -07:00
"audio_network_adaptor/bitrate_controller.cc",
"audio_network_adaptor/bitrate_controller.h",
Adding ChannelController to audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2319873002 Cr-Commit-Position: refs/heads/master@{#14216}
2016-09-14 06:47:36 -07:00
"audio_network_adaptor/channel_controller.cc",
"audio_network_adaptor/channel_controller.h",
Adding basic implementation of AudioNetworkAdaptor. The basic implementation of AudioNetworkAdaptor include the introduction of 1. Controller 2. ControllerManager ControllerManager is to hold all needed controllers. It also orders them according to their significance in dealing with current network condition. Controller provides an interface MakeDecision, which has to be implemented by specific controllers. AudioNetworkAdaptorImpl calls MakeDecision of the controllers in the order decided by ControllerManager to collect EncoderRuntimeConfig. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2306083002 Cr-Commit-Position: refs/heads/master@{#14201}
2016-09-13 13:34:15 -07:00
"audio_network_adaptor/controller.cc",
"audio_network_adaptor/controller.h",
"audio_network_adaptor/controller_manager.cc",
"audio_network_adaptor/controller_manager.h",
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
"audio_network_adaptor/debug_dump_writer.cc",
"audio_network_adaptor/debug_dump_writer.h",
Adding DTX controller to audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2320093002 Cr-Commit-Position: refs/heads/master@{#14260}
2016-09-16 05:54:39 -07:00
"audio_network_adaptor/dtx_controller.cc",
"audio_network_adaptor/dtx_controller.h",
Reland of "Log audio network adapter decisions in event log." This was originally reviewed https://codereview.webrtc.org/2559953002/ It was reverted due to a bug in the original CL, see https://codereview.webrtc.org/2631703002/ This CL is to fix and reland. BUG=webrtc:6845 Review-Url: https://codereview.webrtc.org/2644863002 Cr-Commit-Position: refs/heads/master@{#16242}
2017-01-24 04:54:59 -08:00
"audio_network_adaptor/event_log_writer.cc",
"audio_network_adaptor/event_log_writer.h",
Introduce FecControllerRplrBased 1. Rename FecController to FecControllerPlrBased. 2. Introduce FecControllerRplrBased - a version of FecController that makes its decision based on RPLR instead of PLR. BUG=webrtc:7058 Review-Url: https://codereview.webrtc.org/2672933003 Cr-Commit-Position: refs/heads/master@{#17373}
2017-03-24 04:12:56 -07:00
"audio_network_adaptor/fec_controller_plr_based.cc",
"audio_network_adaptor/fec_controller_plr_based.h",
Adding FrameLengthController to audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2335163002 Cr-Commit-Position: refs/heads/master@{#14339}
2016-09-21 16:00:31 -07:00
"audio_network_adaptor/frame_length_controller.cc",
"audio_network_adaptor/frame_length_controller.h",
Adding AudioNetworkAdaptor interfaces. AudioNetworkAdaptor is supposed to facilitate AudioEncoder to adapt to varying network conditions. This is the first of a sequence of CLs that are to add one implementation of AudioNetworkAdaptor. This CL illustrates the interfaces of the AudioNetworkAdaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2308573002 Cr-Commit-Position: refs/heads/master@{#14115}
2016-09-07 13:51:51 -07:00
"audio_network_adaptor/include/audio_network_adaptor.h",
Avoid code duplication between PLR/RPLR-based FecController BUG=webrtc:7058 Review-Url: https://codereview.webrtc.org/2688613003 Cr-Commit-Position: refs/heads/master@{#17437}
2017-03-29 03:16:58 -07:00
"audio_network_adaptor/util/threshold_curve.h",
Adding AudioNetworkAdaptor interfaces. AudioNetworkAdaptor is supposed to facilitate AudioEncoder to adapt to varying network conditions. This is the first of a sequence of CLs that are to add one implementation of AudioNetworkAdaptor. This CL illustrates the interfaces of the AudioNetworkAdaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2308573002 Cr-Commit-Position: refs/heads/master@{#14115}
2016-09-07 13:51:51 -07:00
]
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
Break the ANA build-target into ANA and ANA-config This is done to solve a dependency-cycle with the RtcEventLog - now the RtcEventLog can depend on the config part of ANA, and be able to peer inside, while the implementation part of ANA can invoke the RtcEventLog. BUG=webrtc:8111 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/3010343002 Cr-Commit-Position: refs/heads/master@{#19793}
2017-09-12 04:38:25 -07:00
public_deps = [
":audio_network_adaptor_config",
]
Adding audio network adaptor to AudioEncoderOpus. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2362703002 Cr-Commit-Position: refs/heads/master@{#14555}
2016-10-06 07:13:54 -07:00
deps = [
Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. This CL only wires up the new stats but does not set the values yet. This will be added in a follow-up CL. BUG=webrtc:8127 Review-Url: https://codereview.webrtc.org/3011623002 Cr-Commit-Position: refs/heads/master@{#19751}
2017-09-08 08:13:19 -07:00
"../../api/audio_codecs:audio_codecs_api",
Delete deprecated rtc_event_log header Bug: webrtc:10206 Change-Id: I9ed3148843c647372993729b87c0e74741ab540b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 12:24:53 +02:00
"../../api/rtc_event_log",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../common_audio",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../../logging:rtc_event_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Move rtc_base/ignore_wundef.h to its own target. Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to move it to its own target and allow its dependant to avoid to take a dependency rtc_base:on rtc_base_approved. Bug: webrtc:9419 Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29548}
2019-10-19 12:03:23 +02:00
"../../rtc_base:ignore_wundef",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
Move file_wrapper.h to rtc_base/system/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 Change-Id: I440974da4d347b09ff042478720d7983056b62b9 Reviewed-on: https://webrtc-review.googlesource.com/21226 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22579}
2018-03-23 10:39:34 +01:00
"../../rtc_base/system:file_wrapper",
Adding audio network adaptor to AudioEncoderOpus. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2362703002 Cr-Commit-Position: refs/heads/master@{#14555}
2016-10-06 07:13:54 -07:00
"../../system_wrappers",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:field_trial",
Replace rtc::Optional with absl::optional in modules/audio_coding This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_coding' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8 Reviewed-on: https://webrtc-review.googlesource.com/84130 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 13:26:36 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Adding audio network adaptor to AudioEncoderOpus. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2362703002 Cr-Commit-Position: refs/heads/master@{#14555}
2016-10-06 07:13:54 -07:00
]
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
if (rtc_enable_protobuf) {
Adding audio network adaptor to AudioEncoderOpus. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2362703002 Cr-Commit-Position: refs/heads/master@{#14555}
2016-10-06 07:13:54 -07:00
deps += [
Creating controller manager from config string in audio network adaptor. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2364403004 Cr-Commit-Position: refs/heads/master@{#14466}
2016-10-02 14:53:37 -07:00
":ana_config_proto",
Relanding of "Adding debug dump to audio network adaptor." The original CL was https://codereview.webrtc.org/2356763002 but got reverted https://codereview.webrtc.org/2362003002/. The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests. BUG=webrtc:6303 Review-Url: https://codereview.webrtc.org/2365723002 Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-22 22:23:20 -07:00
":ana_debug_dump_proto",
]
}
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("neteq") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
"neteq/accelerate.cc",
"neteq/accelerate.h",
"neteq/audio_multi_vector.cc",
"neteq/audio_multi_vector.h",
"neteq/audio_vector.cc",
"neteq/audio_vector.h",
"neteq/background_noise.cc",
"neteq/background_noise.h",
"neteq/buffer_level_filter.cc",
"neteq/buffer_level_filter.h",
"neteq/comfort_noise.cc",
"neteq/comfort_noise.h",
Reland "Avoiding overflow in cross correlation in NetEq." The original CL is https://codereview.webrtc.org/1908623002/ An error was caused by that and this CL fix that problem and reland the CL. BUG= Review-Url: https://codereview.webrtc.org/1931933004 Cr-Commit-Position: refs/heads/master@{#12589}
2016-05-02 01:50:30 -07:00
"neteq/cross_correlation.cc",
"neteq/cross_correlation.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"neteq/decision_logic.cc",
"neteq/decision_logic.h",
"neteq/decoder_database.cc",
"neteq/decoder_database.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",
Add new histograms WebRTC.Audio.(Speech)ExpandRatePercent These two new histograms relate to the packet-loss concealment that happens when audio packets are lost or late for decoding, and the NetEq must resort to extrapolating audio from the previously decoded data. Bug: webrtc:9126 Change-Id: I99cc97e653169fb742da0092653ab99fd10e5d7b Reviewed-on: https://webrtc-review.googlesource.com/67861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22812}
2018-04-10 15:10:26 +02:00
"neteq/expand_uma_logger.cc",
"neteq/expand_uma_logger.h",
Refactor DelayManager into separate Histogram class and make it injectable for testing purposes. Change-Id: I98aa3f992169e598fc1a3dd850400183395fe1fe Bug: webrtc:10333 Reviewed-on: https://webrtc-review.googlesource.com/c/123445 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26797}
2019-02-21 15:42:31 +01:00
"neteq/histogram.cc",
"neteq/histogram.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",
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",
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",
"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",
Hide the AudioEncoderCng class behind a create function And put codecs/cng/webrtc_cng.h in a non-public build target while we're at it. Bug: webrtc:8396 Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c Reviewed-on: https://webrtc-review.googlesource.com/c/108782 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25486}
2018-11-01 11:13:44 +01:00
":webrtc_cng",
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",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"..:module_api_public",
Let NetEq use the PLC output from a decoder This change enables NetEq to use the packet concealment audio (aka PLC) produced by a decoder. The change also includes a new API to the AudioDecoder interface, which lets the decoder implementation generate and deliver concealment audio. Bug: webrtc:9180 Change-Id: Icaacebccf645d4694b0d2d6310f6f2c7132881c4 Reviewed-on: https://webrtc-review.googlesource.com/96340 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24738}
2018-09-05 18:14:52 +02:00
"../../api:array_view",
Reland "Move api/rtp_headers.h to its own build target." This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e Original change's description: > Move api/rtp_headers.h to its own build target. > > Reduces dependencies on the libjingle_peerconnection_api target from > lower-level code. > > Bug: None > Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27213} Bug: None Tbr: kwiberg@webrtc.org Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-20 08:40:23 +01:00
"../../api:rtp_headers",
Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" This reverts commit fab3460a821abe336ab610c6d6dfc0d392dac263. Reason for revert: fix downstream instead Original change's description: > Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" > > This reverts commit 9973933d2e606d64fcdc753acb9ba3afd6e30569. > > Reason for revert: breaking downstream projects and not reviewed by direct owners > > Original change's description: > > Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > This reverts commit 24192c267a40eb7d6b1850489ccdbf7a84f8ff0f. > > > > Reason for revert: Analyzed the performance regression in more detail. > > > > Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit 1796a820f60cb9429bf4bcf13a40a41794ac8fb0 has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac. > > > > There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall. > > > > Original change's description: > > > Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > > > This reverts commit 3e8ef940fe86cf6285afb80e68d2a0bedc631b9f. > > > > > > Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260. > > > > > > Original change's description: > > > > Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. > > > > > > > > This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time. > > > > > > > > Bug: webrtc:10668 > > > > Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890 > > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > > Reviewed-by: Minyue Li <minyue@webrtc.org> > > > > Commit-Queue: Chen Xing <chxg@google.com> > > > > Cr-Commit-Position: refs/heads/master@{#28434} > > > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > > > Bug: webrtc:10668, chromium:982260 > > > Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339 > > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28561} > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:10668, chromium:982260 > > Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707 > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Commit-Queue: Chen Xing <chxg@google.com> > > Cr-Commit-Position: refs/heads/master@{#28664} > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10668, chromium:982260 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28671} TBR=alessiob@webrtc.org,kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com Change-Id: Id43b7b3da79b4f48004b41767482bae1c1fa1e16 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10668, chromium:982260 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146713 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28672}
2019-07-24 16:47:02 +00:00
"../../api:rtp_packet_info",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_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",
Move NetEq headers to api/ This CL also introduces NetEqFactory and NetEqControllerFactory interfaces, as well as several convenience classes for working with them: DefaultNetEqFactory, DefaultNetEqControllerFactory and CustomNetEqFactory. Bug: webrtc:11005 Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29671}
2019-10-31 14:38:11 +01:00
"../../api/neteq:neteq_api",
"../../api/neteq:neteq_controller_api",
"../../api/neteq:tick_timer",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
This CL removes all usages of our custom ostream << overloads. This prepares us for removing them altogether. Bug: webrtc:8982 Change-Id: I66002cc8d4bf0e07925766d568d2498422f0f38e Reviewed-on: https://webrtc-review.googlesource.com/64142 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Henrik Grunell <henrikg@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22707}
2018-04-03 13:40:05 +02:00
"../../rtc_base:audio_format_to_string",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:gtest_prod",
"../../rtc_base:rtc_base_approved",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../../rtc_base:safe_minmax",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:sanitizer",
Improve buffer level estimation with DTX and add CNG time stretching. The functionality is hidden behind field trial for experimentation. Bug: webrtc:10736 Change-Id: I1daf60966717c3ea43bf6ee16d190290ab740ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144059 Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28474}
2019-07-03 16:00:30 +02:00
"../../rtc_base/experiments:field_trial_parser",
Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix Bug: chromium:805946 Change-Id: Ibb5dce9af27d0e48c9aee6b0a860b6f62b3c76a0 Reviewed-on: https://webrtc-review.googlesource.com/46145 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21889}
2018-02-05 10:33:35 +01:00
"../../rtc_base/system:fallthrough",
Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" This reverts commit fab3460a821abe336ab610c6d6dfc0d392dac263. Reason for revert: fix downstream instead Original change's description: > Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" > > This reverts commit 9973933d2e606d64fcdc753acb9ba3afd6e30569. > > Reason for revert: breaking downstream projects and not reviewed by direct owners > > Original change's description: > > Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > This reverts commit 24192c267a40eb7d6b1850489ccdbf7a84f8ff0f. > > > > Reason for revert: Analyzed the performance regression in more detail. > > > > Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit 1796a820f60cb9429bf4bcf13a40a41794ac8fb0 has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac. > > > > There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall. > > > > Original change's description: > > > Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > > > This reverts commit 3e8ef940fe86cf6285afb80e68d2a0bedc631b9f. > > > > > > Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260. > > > > > > Original change's description: > > > > Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. > > > > > > > > This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time. > > > > > > > > Bug: webrtc:10668 > > > > Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890 > > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > > Reviewed-by: Minyue Li <minyue@webrtc.org> > > > > Commit-Queue: Chen Xing <chxg@google.com> > > > > Cr-Commit-Position: refs/heads/master@{#28434} > > > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > > > Bug: webrtc:10668, chromium:982260 > > > Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339 > > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28561} > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:10668, chromium:982260 > > Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707 > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Commit-Queue: Chen Xing <chxg@google.com> > > Cr-Commit-Position: refs/heads/master@{#28664} > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10668, chromium:982260 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28671} TBR=alessiob@webrtc.org,kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com Change-Id: Id43b7b3da79b4f48004b41767482bae1c1fa1e16 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10668, chromium:982260 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146713 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28672}
2019-07-24 16:47:02 +00:00
"../../system_wrappers",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. Bug: webrtc:5876 Change-Id: Ica2d47ca45b8ef01a548d8dbe31dbed740a0ebda Reviewed-on: https://webrtc-review.googlesource.com/c/106820 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25306}
2018-10-22 09:48:08 +02:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional in modules/audio_coding This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_coding' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8 Reviewed-on: https://webrtc-review.googlesource.com/84130 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 13:26:36 +02:00
"//third_party/abseil-cpp/absl/types:optional",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
# Although providing only test support, this target must be outside of the
# rtc_include_tests conditional. The reason is that it supports fuzzer tests
# that ultimately are built and run as a part of the Chromium ecosystem, which
# does not set the rtc_include_tests flag.
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("neteq_tools_minimal") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
sources = [
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
"neteq/tools/audio_sink.cc",
"neteq/tools/audio_sink.h",
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
"neteq/tools/encode_neteq_input.cc",
"neteq/tools/encode_neteq_input.h",
Add NetEqInput::PacketData::ToString method This new method prints information about the packet. BUG=webrtc:7467 Review-Url: https://codereview.webrtc.org/2844283002 Cr-Commit-Position: refs/heads/master@{#17922}
2017-04-28 01:35:53 -07:00
"neteq/tools/neteq_input.cc",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
"neteq/tools/neteq_input.h",
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
"neteq/tools/neteq_test.cc",
"neteq/tools/neteq_test.h",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
"neteq/tools/packet.cc",
"neteq/tools/packet.h",
"neteq/tools/packet_source.cc",
"neteq/tools/packet_source.h",
Relanding "Setting up an RTP input fuzzer for NetEq" The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
2016-10-05 02:27:42 -07:00
]
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
deps = [
":neteq",
Restructure neteq_rtpplay into a library with small executable wrapper. Most of the code in neteq_rtpplay is moved into a factory class for NetEqTest. The factory method takes the same argc and argv arguments as neteq_rtpplay. This CL also adds a small public API for neteq_test to allow easy integration into external software. Bug: webrtc:9667 Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d Reviewed-on: https://webrtc-review.googlesource.com/96100 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24531}
2018-09-03 11:49:27 +02:00
"../../api:neteq_simulator_api",
Reland "Move api/rtp_headers.h to its own build target." This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e Original change's description: > Move api/rtp_headers.h to its own build target. > > Reduces dependencies on the libjingle_peerconnection_api target from > lower-level code. > > Bug: None > Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27213} Bug: None Tbr: kwiberg@webrtc.org Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-20 08:40:23 +01:00
"../../api:rtp_headers",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../../api/audio_codecs:audio_codecs_api",
Move NetEq headers to api/ This CL also introduces NetEqFactory and NetEqControllerFactory interfaces, as well as several convenience classes for working with them: DefaultNetEqFactory, DefaultNetEqControllerFactory and CustomNetEqFactory. Bug: webrtc:11005 Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29671}
2019-10-31 14:38:11 +01:00
"../../api/neteq:custom_neteq_factory",
"../../api/neteq:default_neteq_controller_factory",
"../../api/neteq:neteq_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" This reverts commit fab3460a821abe336ab610c6d6dfc0d392dac263. Reason for revert: fix downstream instead Original change's description: > Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" > > This reverts commit 9973933d2e606d64fcdc753acb9ba3afd6e30569. > > Reason for revert: breaking downstream projects and not reviewed by direct owners > > Original change's description: > > Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > This reverts commit 24192c267a40eb7d6b1850489ccdbf7a84f8ff0f. > > > > Reason for revert: Analyzed the performance regression in more detail. > > > > Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit 1796a820f60cb9429bf4bcf13a40a41794ac8fb0 has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac. > > > > There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall. > > > > Original change's description: > > > Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > > > This reverts commit 3e8ef940fe86cf6285afb80e68d2a0bedc631b9f. > > > > > > Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260. > > > > > > Original change's description: > > > > Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. > > > > > > > > This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time. > > > > > > > > Bug: webrtc:10668 > > > > Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890 > > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > > Reviewed-by: Minyue Li <minyue@webrtc.org> > > > > Commit-Queue: Chen Xing <chxg@google.com> > > > > Cr-Commit-Position: refs/heads/master@{#28434} > > > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > > > Bug: webrtc:10668, chromium:982260 > > > Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339 > > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28561} > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:10668, chromium:982260 > > Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707 > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Commit-Queue: Chen Xing <chxg@google.com> > > Cr-Commit-Position: refs/heads/master@{#28664} > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10668, chromium:982260 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28671} TBR=alessiob@webrtc.org,kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com Change-Id: Id43b7b3da79b4f48004b41767482bae1c1fa1e16 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10668, chromium:982260 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146713 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28672}
2019-07-24 16:47:02 +00:00
"../../system_wrappers",
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",
Added new feature to print a text log to neteq_rtpplay This will print out the major events during a NetEq simulation. Bug: b/116685514 Change-Id: Iab172e9a9115695b42c67628d5523c727359bb89 Reviewed-on: https://webrtc-review.googlesource.com/c/114320 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26019}
2018-12-14 16:49:12 +01:00
"../rtp_rtcp:rtp_rtcp_format",
Replace rtc::Optional with absl::optional in modules/audio_coding This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_coding' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8 Reviewed-on: https://webrtc-review.googlesource.com/84130 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 13:26:36 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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
]
Add missing conditional defines to neteq test and tools targets The .cc source files listed below #ifdef for WEBRTC_CODEC_OPUS and WEBRTC_CODEC_ILBC but the build files don't include the defines. modules/audio_coding/neteq/tools/neteq_test.cc modules/audio_coding/neteq/tools/neteq_test_factory.cc Bug: None Change-Id: I6065021f68e58d0e5663acd006a9865bf265adc0 Reviewed-on: https://webrtc-review.googlesource.com/c/109925 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25555}
2018-11-08 09:46:10 +01:00
defines = audio_codec_defines
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("neteq_test_tools") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
testonly = true
sources = [
"neteq/tools/audio_checksum.h",
"neteq/tools/audio_loop.cc",
"neteq/tools/audio_loop.h",
"neteq/tools/constant_pcm_packet_source.cc",
"neteq/tools/constant_pcm_packet_source.h",
Added support for skipping get_audio events, adding dummy packets and setting a field trial string. Bug: webrtc:10337 Change-Id: I0507da4d955daa914af774c946be16a4168be21a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150780 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29392}
2019-10-07 13:18:18 +02:00
"neteq/tools/initial_packet_inserter_neteq_input.cc",
"neteq/tools/initial_packet_inserter_neteq_input.h",
Break out NetEqEventLogInput to separate source files Building NetEqEventLogInput requires protobuf support, while building NetEqRtpDumpInput located in the same file does not. This makes both classes unusable when protobuf support is lacking. With this CL, the NetEqEventLogInput is broken out into separate files, to allow usage of NetEqRtpDumpInput even when protobufs are not supported. Bug: webrtc:9421 Change-Id: I55efec4ec259713654566cdaa00d2e34c5e9a60f Reviewed-on: https://webrtc-review.googlesource.com/84587 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23803}
2018-07-02 14:05:37 +02:00
"neteq/tools/neteq_packet_source_input.cc",
"neteq/tools/neteq_packet_source_input.h",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"neteq/tools/output_audio_file.h",
"neteq/tools/output_wav_file.h",
"neteq/tools/rtp_file_source.cc",
"neteq/tools/rtp_file_source.h",
"neteq/tools/rtp_generator.cc",
"neteq/tools/rtp_generator.h",
]
deps = [
":pcm16b",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Reland "Move api/rtp_headers.h to its own build target." This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e Original change's description: > Move api/rtp_headers.h to its own build target. > > Reduces dependencies on the libjingle_peerconnection_api target from > lower-level code. > > Bug: None > Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27213} Bug: None Tbr: kwiberg@webrtc.org Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-20 08:40:23 +01:00
"../../api:rtp_headers",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../../common_audio",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../rtc_base",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../../test:rtp_test_utils",
"../rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
RtcEventLogSource no longer uses deprecated parsing functions. Also remove header extension map from NetEqEventLogInput and RtcEventLogSource. Bug: webrtc:8111 Change-Id: Ic9be7b03e32ab8aa12284596e21e53b6763f483a Reviewed-on: https://webrtc-review.googlesource.com/c/102622 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25122}
2018-10-11 16:51:23 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
public_deps = [
":neteq_tools",
":neteq_tools_minimal",
]
if (rtc_enable_protobuf) {
sources += [
Break out NetEqEventLogInput to separate source files Building NetEqEventLogInput requires protobuf support, while building NetEqRtpDumpInput located in the same file does not. This makes both classes unusable when protobuf support is lacking. With this CL, the NetEqEventLogInput is broken out into separate files, to allow usage of NetEqRtpDumpInput even when protobufs are not supported. Bug: webrtc:9421 Change-Id: I55efec4ec259713654566cdaa00d2e34c5e9a60f Reviewed-on: https://webrtc-review.googlesource.com/84587 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23803}
2018-07-02 14:05:37 +02:00
"neteq/tools/neteq_event_log_input.cc",
"neteq/tools/neteq_event_log_input.h",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
deps += [ ":rtc_event_log_source" ]
}
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("neteq_tools") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
sources = [
"neteq/tools/fake_decode_from_file.cc",
"neteq/tools/fake_decode_from_file.h",
Add NetEq delay plotting to event_log_visualizer This CL adds the capability to analyze and plot how NetEq behaves in response to a network trace. BUG=webrtc:7467 Review-Url: https://codereview.webrtc.org/2876423002 Cr-Commit-Position: refs/heads/master@{#18590}
2017-06-14 06:09:58 -07:00
"neteq/tools/neteq_delay_analyzer.cc",
"neteq/tools/neteq_delay_analyzer.h",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"neteq/tools/neteq_replacement_input.cc",
"neteq/tools/neteq_replacement_input.h",
Seperate NetEq stats getter to use in other tools. Bug: webrtc:9147 Change-Id: I251618bbb542d89b3d38c3ea424b1e55c0a5f2b2 Reviewed-on: https://webrtc-review.googlesource.com/69806 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22971}
2018-04-16 14:33:53 +02:00
"neteq/tools/neteq_stats_getter.cc",
"neteq/tools/neteq_stats_getter.h",
Restructure neteq_rtpplay into a library with small executable wrapper. Most of the code in neteq_rtpplay is moved into a factory class for NetEqTest. The factory method takes the same argc and argv arguments as neteq_rtpplay. This CL also adds a small public API for neteq_test to allow easy integration into external software. Bug: webrtc:9667 Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d Reviewed-on: https://webrtc-review.googlesource.com/96100 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24531}
2018-09-03 11:49:27 +02:00
"neteq/tools/neteq_stats_plotter.cc",
"neteq/tools/neteq_stats_plotter.h",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
deps = [
Include module_common_types.h only where needed Bug: None Change-Id: I73d493f8f186b429c7be808f4dfac0398f150931 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153891 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29277}
2019-09-23 10:31:16 +02:00
"..:module_api_public",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../../api/audio_codecs:audio_codecs_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional in modules/audio_coding This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_coding' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8 Reviewed-on: https://webrtc-review.googlesource.com/84130 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 13:26:36 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
public_deps = [
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
":neteq_input_audio_tools",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
":neteq_tools_minimal",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("neteq_input_audio_tools") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
sources = [
"neteq/tools/input_audio_file.cc",
"neteq/tools/input_audio_file.h",
"neteq/tools/resample_input_audio_file.cc",
"neteq/tools/resample_input_audio_file.h",
]
deps = [
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
"../../rtc_base:rtc_base_approved",
]
}
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
if (rtc_enable_protobuf) {
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("rtc_event_log_source") {
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
testonly = true
sources = [
"neteq/tools/rtc_event_log_source.cc",
"neteq/tools/rtc_event_log_source.h",
]
deps = [
Fix circular dependency in rtc_event_log. Bug: webrtc:6828 Change-Id: Ief948b6799455cfda6cb89e2e632f5fd42df0881 Reviewed-on: https://webrtc-review.googlesource.com/25840 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20928}
2017-11-24 13:51:52 +01:00
":neteq_tools_minimal",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
"../../logging:rtc_event_log_parser",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Stop using public_deps in logging/. Bug: webrtc:8603 Change-Id: Id0df997620a27e47067e4b21e4e8db16aec90640 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30940 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21228}
2017-12-12 12:46:13 +01:00
"../rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
RtcEventLogSource no longer uses deprecated parsing functions. Also remove header extension map from NetEqEventLogInput and RtcEventLogSource. Bug: webrtc:8111 Change-Id: Ic9be7b03e32ab8aa12284596e21e53b6763f483a Reviewed-on: https://webrtc-review.googlesource.com/c/102622 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25122}
2018-10-11 16:51:23 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
public_deps = [
"../../logging:rtc_event_log_proto",
]
}
}
Expose OpusTest class for re-use as chromium test. This CL allows to trigger related tests when rolling opus (at chromium side). Bug: chromium:1002973 Change-Id: I811d17233367cabc8b4aa8ab5bbf3e92359afbce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158887 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29716}
2019-11-06 21:04:55 +01:00
# Allow to re-use some test classes from chromium.
rtc_library("audio_coding_modules_tests_shared") {
testonly = true
visibility = []
visibility = [ "*" ]
sources = [
"test/PCMFile.cc",
"test/PCMFile.h",
"test/TestStereo.cc",
"test/TestStereo.h",
"test/opus_test.cc",
"test/opus_test.h",
]
deps = [
":audio_coding",
":audio_coding_module_typedefs",
":webrtc_opus_wrapper",
"..:module_api",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/audio_codecs:builtin_audio_encoder_factory",
"../../rtc_base:checks",
"../../rtc_base:stringutils",
"../../test:fileutils",
"../../test:test_support",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
defines = audio_coding_defines
}
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) {
AudioCodingModule: Remove support for creating encoders After this CL, all audio encoders have to be injected by the caller. This means that there is no special "built-in" set of codecs, and users won't have to pay the binary size and security costs of codecs they aren't using. Bug: webrtc:8396 Change-Id: Idb0959ce395940c8bb3bbb49256cdcd84fc87bb6 Reviewed-on: https://webrtc-review.googlesource.com/c/103821 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25600}
2018-11-12 14:21:58 +01:00
audio_coding_deps = [
"../../common_audio",
"../../system_wrappers",
":audio_encoder_cng",
":g711",
":g722",
":pcm16b",
]
if (rtc_include_ilbc) {
audio_coding_deps += [ ":ilbc" ]
}
if (rtc_include_opus) {
audio_coding_deps += [ ":webrtc_opus" ]
}
if (current_cpu == "arm") {
audio_coding_deps += [ ":isac_fix" ]
} else {
audio_coding_deps += [ ":isac" ]
}
if (!build_with_mozilla && !build_with_chromium) {
audio_coding_deps += [ ":red" ]
}
Add missing files to audio_coding Bug: webrtc:7650 Change-Id: I8d7d8c3998799404ec6283896883d195468cdfdc Reviewed-on: https://webrtc-review.googlesource.com/37622 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21502}
2018-01-05 11:46:26 +01:00
rtc_source_set("mocks") {
testonly = true
sources = [
"audio_network_adaptor/mock/mock_audio_network_adaptor.h",
"audio_network_adaptor/mock/mock_controller.h",
"audio_network_adaptor/mock/mock_controller_manager.h",
"audio_network_adaptor/mock/mock_debug_dump_writer.h",
]
deps = [
":audio_network_adaptor",
"../../test:test_support",
]
}
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
group("audio_coding_tests") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
testonly = true
public_deps = [
":acm_receive_test",
":acm_send_test",
":audio_codec_speed_tests",
":audio_decoder_unittests",
":audio_decoder_unittests",
":g711_test",
":g722_test",
":ilbc_test",
":isac_api_test",
":isac_fix_test",
":isac_switch_samprate_test",
":isac_test",
":neteq_ilbc_quality_test",
":neteq_isac_quality_test",
":neteq_opus_quality_test",
Add pcm16b quality test supporting 48khz. Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Bug: webrtc:8836 Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Reviewed-on: https://webrtc-review.googlesource.com/47400 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21878}
2018-02-02 17:21:46 +01:00
":neteq_pcm16b_quality_test",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
":neteq_pcmu_quality_test",
":neteq_speed_test",
":rtp_analyze",
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
":rtp_encode",
Replacing the legacy tool RTPjitter with a new rtp_jitter This new tool provides the similar functionality as the legacy tool, but is implemented using less legacy helpers. It also replaces RTPtimeshift and RTPchange. The most significant change versus the old RTPjitter tool is that the new tool takes the timing data in the form of integers in a text file (instead of the binary data file used by the old tool). This should make it easier to create custom timing files when needed. Bug: webrtc:2692 Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287 Reviewed-on: https://webrtc-review.googlesource.com/25700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:39:39 +01:00
":rtp_jitter",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
":rtpcat",
":webrtc_opus_fec_test",
]
if (rtc_enable_protobuf) {
public_deps += [ ":neteq_rtpplay" ]
}
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_coding_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
testonly = true
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
sources = [
"test/Channel.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/Channel.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/EncodeDecodeTest.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/EncodeDecodeTest.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/PacketLossTest.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/PacketLossTest.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/RTPFile.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/RTPFile.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/TestAllCodecs.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/TestAllCodecs.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/TestRedFec.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/TestRedFec.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/TestVADDTX.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/TestVADDTX.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/Tester.cc",
"test/TwoWayCommunication.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/TwoWayCommunication.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/iSACTest.cc",
Add untracked header files to GN targets in audio_coding BUG=webrtc:7619 Review-Url: https://codereview.webrtc.org/2875033002 Cr-Commit-Position: refs/heads/master@{#18101}
2017-05-11 07:18:06 -07:00
"test/iSACTest.h",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"test/target_delay_unittest.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",
Expose OpusTest class for re-use as chromium test. This CL allows to trigger related tests when rolling opus (at chromium side). Bug: chromium:1002973 Change-Id: I811d17233367cabc8b4aa8ab5bbf3e92359afbce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158887 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29716}
2019-11-06 21:04:55 +01:00
":audio_coding_modules_tests_shared",
Hide the AudioEncoderCng class behind a create function And put codecs/cng/webrtc_cng.h in a non-public build target while we're at it. Bug: webrtc:8396 Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c Reviewed-on: https://webrtc-review.googlesource.com/c/108782 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25486}
2018-11-01 11:13:44 +01:00
":audio_encoder_cng",
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
":pcm16b_c",
AudioCodingModuleTest.TestRedFec: Don't let the ACM create audio encoders It will soon lose the ability to do so. Bug: webrtc:8396 Change-Id: If120afa37325c00ae2c3e9a9bd75bf89c8897f8c Reviewed-on: https://webrtc-review.googlesource.com/c/103441 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24979}
2018-10-04 12:38:03 +02:00
":red",
Renaming opus_interface.c to opus_interface.cc. This is to allow advanced features of WebRTC/Chrome e.g., field trials. More style compliant changes may follow up. Only a minimal (not in terms of line changes) is applied, so that presubmit does not complain. These changes include 1. removing unused headers. 2. eliminating c-style casting. Bug: b/143582588 Change-Id: I6d0fd926c542ab0afdc38cc4bf03aaf584ec13dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158670 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29657}
2019-10-29 21:36:13 +01:00
":webrtc_opus_wrapper",
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",
Include module_common_types.h only where needed Bug: None Change-Id: I73d493f8f186b429c7be808f4dfac0398f150931 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153891 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29277}
2019-09-23 10:31:16 +02:00
"../../api:rtp_headers",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
AudioCodingModuleTest.TwoWayCommunication: Don't let the ACM create encoders It will soon lose the ability to do so. Bug: webrtc:8396 Change-Id: I7d8e1549c44628fc9bdf2480468a0f1d3ae812f2 Reviewed-on: https://webrtc-review.googlesource.com/102062 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24853}
2018-09-26 11:22:50 +02:00
"../../api/audio_codecs:audio_codecs_api",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:builtin_audio_decoder_factory",
AudioCodingModuleTest.TestAllCodecs: Create audio encoders the new way Specifically, don't expect the ACM to be able to create encoders; we have to give it an encoder that we make ourselves. To make it work, I had to add support for the "ptime" parameter to the L16 codec. Bug: webrtc:8396 Change-Id: I3869422882611d2eed65d6c849ea7cd3ad6bd126 Reviewed-on: https://webrtc-review.googlesource.com/87423 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24217}
2018-07-06 15:40:14 +02:00
"../../api/audio_codecs:builtin_audio_encoder_factory",
AudioCodingModuleTest.TestRedFec: Don't let the ACM create audio encoders It will soon lose the ability to do so. Bug: webrtc:8396 Change-Id: If120afa37325c00ae2c3e9a9bd75bf89c8897f8c Reviewed-on: https://webrtc-review.googlesource.com/c/103441 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24979}
2018-10-04 12:38:03 +02:00
"../../api/audio_codecs/L16:audio_decoder_L16",
"../../api/audio_codecs/L16:audio_encoder_L16",
"../../api/audio_codecs/g711:audio_decoder_g711",
"../../api/audio_codecs/g711:audio_encoder_g711",
"../../api/audio_codecs/g722:audio_decoder_g722",
"../../api/audio_codecs/g722:audio_encoder_g722",
VAD/DTX tests: Don't let the ACM create audio encoders It will soon lose the ability to do so. Bug: webrtc:8396 Change-Id: I06dce417bba855b57130bd1a052988b2f235dcbd Reviewed-on: https://webrtc-review.googlesource.com/102882 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24921}
2018-10-01 17:26:11 +02:00
"../../api/audio_codecs/ilbc:audio_decoder_ilbc",
"../../api/audio_codecs/ilbc:audio_encoder_ilbc",
"../../api/audio_codecs/isac:audio_decoder_isac_float",
AudioCodingModuleTest.TestIsac: Don't rely on the ACM to create encoders It will soon lose the ability to do so. Bug: webrtc:8396 Change-Id: Id413204e53afec28495dff0873f027a56caed80f Reviewed-on: https://webrtc-review.googlesource.com/101861 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24834}
2018-09-25 14:48:33 +02:00
"../../api/audio_codecs/isac:audio_encoder_isac_float",
VAD/DTX tests: Don't let the ACM create audio encoders It will soon lose the ability to do so. Bug: webrtc:8396 Change-Id: I06dce417bba855b57130bd1a052988b2f235dcbd Reviewed-on: https://webrtc-review.googlesource.com/102882 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24921}
2018-10-01 17:26:11 +02:00
"../../api/audio_codecs/opus:audio_decoder_opus",
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../common_audio",
Remove incompatiblities with absl::optional in audio_coding PCMFile.cc uses RTC_DCHECK. include and depend on rtc_base:checks target directly change usage of value_or by using explicit constructor instead of implicit Bug: webrtc:9078 Change-Id: I63c596b8a05b387e56df846b15c33a605fbad4e6 Reviewed-on: https://webrtc-review.googlesource.com/69985 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22897}
2018-04-17 10:10:57 +02:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Move rw_lock_wrapper.h to rtc_base/synchronization/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 NOPRESUBMIT=true Change-Id: Ie2879aca5fc1667e4222499d2a8fc2bba9ae2425 Reviewed-on: https://webrtc-review.googlesource.com/21328 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22587}
2018-03-23 14:53:54 +01:00
"../../rtc_base/synchronization:rw_lock_wrapper",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"../../test:test_support",
Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. Bug: webrtc:5876 Change-Id: Ica2d47ca45b8ef01a548d8dbe31dbed740a0ebda Reviewed-on: https://webrtc-review.googlesource.com/c/106820 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25306}
2018-10-22 09:48:08 +02:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional in modules/audio_coding This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_coding' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8 Reviewed-on: https://webrtc-review.googlesource.com/84130 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 13:26:36 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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
]
defines = audio_coding_defines
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_coding_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
testonly = true
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
sources = [
"codecs/opus/opus_complexity_unittest.cc",
"neteq/test/neteq_performance_unittest.cc",
]
deps = [
":neteq_test_support",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_test_tools",
Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix We've done this previously with the other audio encoders, but Opus had to wait until all external users had been updated. BUG=webrtc:7847 Change-Id: I70422d7b6c715f32a43bee88febcf6b6155e18b3 Reviewed-on: https://webrtc-review.googlesource.com/8000 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20424}
2017-10-25 09:57:40 +02:00
"../../api/audio_codecs/opus:audio_encoder_opus",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:field_trial",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Reland "iOS: Save perf results under Documents/perf_result.json" This will require a manual roll to downstream projects, since the //test:perf_test target was introduced. This is a reland of 10a8e7a9b5261a7e3ce19900ba3511be3b5911f8 Original change's description: > iOS: Save perf results under Documents/perf_result.json > > TBR=henrika@webrtc.org > > Bug: webrtc:7156 > Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5 > Reviewed-on: https://webrtc-review.googlesource.com/29202 > Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21244} R=henrika@webrtc.org, phoglund@webrtc.org Bug: webrtc:7156 Change-Id: I85fc7bc5fce0894af90017b71b9952b61b523424 Reviewed-on: https://webrtc-review.googlesource.com/37643 Reviewed-by: Patrik Höglund <phoglund@google.com> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21518}
2018-01-05 15:34:09 +01:00
"../../test:perf_test",
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
"../../test:test_support",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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 + [
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"..:module_api",
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",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_tools",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../test:test_support",
Depend directly on destination targets. Makes 'gn check' happy. Followup to https://webrtc-review.googlesource.com/c/src/+/106820 Bug: webrtc:5876, webrtc:9855 Change-Id: I33fa2c31ba26dc10c9a9c17da0ffed255c1f4d5b Reviewed-on: https://webrtc-review.googlesource.com/c/108760 Commit-Queue: Yves Gerey <yvesg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25447}
2018-10-30 22:23:02 +01:00
"//third_party/abseil-cpp/absl/strings",
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",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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 + [
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
":audio_coding",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_tools",
AudioCodingModule::Create(): Require caller to supply an AudioDecoderFactory So that we don't have to be capable of creating one ourselves, which requires a dependency on the audio decoders. BUG=webrtc:5801, webrtc:8396 Change-Id: I80749ec3b86cba73994307046d05964f59167d44 Reviewed-on: https://webrtc-review.googlesource.com/18440 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22774}
2018-04-06 10:06:42 +02:00
"../../api/audio_codecs:builtin_audio_decoder_factory",
Audio encoder tests: Create audio encoders the new way Specifically, don't expect the ACM to be able to create encoders; we have to give it an encoder that we make ourselves. The new way of creating encoders used a 32 kbit/s bitrate unconditionally for iSAC; I had to change it to 32 kbit/s for 16 kHz and 56 kbit/s for 32 kHz, which is what the old way of creating encoders has used since forever. I also had to change some test expectations on Opus, because the new way defaults to 32 kbit/s for mono and 64 kbit/s for stereo (which I believe to be correct), while the old way defaults to 64 kbit/s in both cases. Bug: webrtc:8396 Change-Id: I3aab944175a8e27f4c63380e822b27e839bba7f2 Reviewed-on: https://webrtc-review.googlesource.com/94540 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24375}
2018-08-16 15:01:12 +02:00
"../../api/audio_codecs:builtin_audio_encoder_factory",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../../api/audio_codecs:audio_codecs_api",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../test:test_support",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"//testing/gtest",
]
}
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
audio_decoder_unittests_resources =
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
[ "../../resources/audio_coding/testfile32kHz.pcm" ]
GN: Fix resource files for iOS test target In order to get resource files to be properly packaged into the .app for a unit test on iOS, the resource files needs to be listed as sources in a bundle_data target. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2292853002 Cr-Commit-Position: refs/heads/master@{#13968}
2016-08-30 02:53:49 -07:00
if (is_ios) {
bundle_data("audio_decoder_unittests_bundle_data") {
testonly = true
sources = audio_decoder_unittests_resources
outputs = [
"{{bundle_resources_dir}}/{{source_file_part}}",
]
}
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_test("audio_decoder_unittests") {
GN conversion of audio_decoder_unittests BUG=webrtc 6036 Review-Url: https://codereview.webrtc.org/2180853006 Cr-Commit-Position: refs/heads/master@{#13588}
2016-08-01 07:49:42 -07:00
testonly = true
sources = [
"neteq/audio_decoder_unittest.cc",
]
defines = neteq_defines
Enable clang::find_bad_constructs for audio_coding (part 1/2). This CL removes //build/config/clang:find_bad_constructs from the suppressed_configs list, which means that clang:find_bad_constructs is now enabled on these translation units. Bug: webrtc:9251, webrtc:163 Change-Id: I6a7d4964723a5e195189aac30a83d9e924e61dd7 Reviewed-on: https://webrtc-review.googlesource.com/89743 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24053}
2018-07-20 13:59:20 +02:00
deps = [
":ilbc",
":isac",
":isac_fix",
":neteq",
":neteq_tools",
"../../test:fileutils",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../common_audio",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
Enable clang::find_bad_constructs for audio_coding (part 1/2). This CL removes //build/config/clang:find_bad_constructs from the suppressed_configs list, which means that clang:find_bad_constructs is now enabled on these translation units. Bug: webrtc:9251, webrtc:163 Change-Id: I6a7d4964723a5e195189aac30a83d9e924e61dd7 Reviewed-on: https://webrtc-review.googlesource.com/89743 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24053}
2018-07-20 13:59:20 +02:00
"../../test:test_main",
"//testing/gtest",
Harmonize paths to dependent targets. This CL consistently use: * relative paths for WebRTC dependent targets (test_support) * absolute paths for shared dependent targets (abseil) This is a necessary (but insufficient) step to build WebRTC tests from Chromium tree (rtc_include_tests=true), since test/ doesn't sit anymore in the top level directory. We also make sure that target declarations and uses are consistent in regard to build_with_chromium flag. Bug: webrtc:9943 Bug: webrtc:9855 Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b Reviewed-on: https://webrtc-review.googlesource.com/c/108720 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25445}
2018-10-30 21:12:42 +01:00
"../../test:test_support",
Enable clang::find_bad_constructs for audio_coding (part 1/2). This CL removes //build/config/clang:find_bad_constructs from the suppressed_configs list, which means that clang:find_bad_constructs is now enabled on these translation units. Bug: webrtc:9251, webrtc:163 Change-Id: I6a7d4964723a5e195189aac30a83d9e924e61dd7 Reviewed-on: https://webrtc-review.googlesource.com/89743 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24053}
2018-07-20 13:59:20 +02:00
] + audio_coding_deps
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: 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
}
Enable clang::find_bad_constructs for audio_coding (part 1/2). This CL removes //build/config/clang:find_bad_constructs from the suppressed_configs list, which means that clang:find_bad_constructs is now enabled on these translation units. Bug: webrtc:9251, webrtc:163 Change-Id: I6a7d4964723a5e195189aac30a83d9e924e61dd7 Reviewed-on: https://webrtc-review.googlesource.com/89743 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24053}
2018-07-20 13:59:20 +02:00
}
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: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (rtc_enable_protobuf) {
proto_library("neteq_unittest_proto") {
sources = [
"neteq/neteq_unittest.proto",
]
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
proto_out_dir = "modules/audio_coding/neteq"
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
GN: Add neteq_rtpplay and rtc_event_log_source BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2081113003 Cr-Commit-Position: refs/heads/master@{#13239}
2016-06-21 05:38:42 -07:00
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("neteq_test_factory") {
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
Restructure neteq_rtpplay into a library with small executable wrapper. Most of the code in neteq_rtpplay is moved into a factory class for NetEqTest. The factory method takes the same argc and argv arguments as neteq_rtpplay. This CL also adds a small public API for neteq_test to allow easy integration into external software. Bug: webrtc:9667 Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d Reviewed-on: https://webrtc-review.googlesource.com/96100 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24531}
2018-09-03 11:49:27 +02:00
visibility += webrtc_default_visibility
Add missing conditional defines to neteq test and tools targets The .cc source files listed below #ifdef for WEBRTC_CODEC_OPUS and WEBRTC_CODEC_ILBC but the build files don't include the defines. modules/audio_coding/neteq/tools/neteq_test.cc modules/audio_coding/neteq/tools/neteq_test_factory.cc Bug: None Change-Id: I6065021f68e58d0e5663acd006a9865bf265adc0 Reviewed-on: https://webrtc-review.googlesource.com/c/109925 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25555}
2018-11-08 09:46:10 +01:00
defines = audio_codec_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 = [
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Move command line flags out of NetEqTestFactory This replaces the use of command-line flags with the use of a config struct. This makes it easier for non command-line applications to use the NetEqTestFactory to run simulations. Bug: webrtc:10337 Change-Id: I24533bf206e70e12db9af8d9675769c1ff7c7d48 Reviewed-on: https://webrtc-review.googlesource.com/c/123600 Reviewed-by: Pablo Barrera González <barrerap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26887}
2019-02-26 13:03:21 +01:00
"//third_party/abseil-cpp/absl/types:optional",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
]
GN: Add neteq_rtpplay and rtc_event_log_source BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2081113003 Cr-Commit-Position: refs/heads/master@{#13239}
2016-06-21 05:38:42 -07:00
sources = [
Restructure neteq_rtpplay into a library with small executable wrapper. Most of the code in neteq_rtpplay is moved into a factory class for NetEqTest. The factory method takes the same argc and argv arguments as neteq_rtpplay. This CL also adds a small public API for neteq_test to allow easy integration into external software. Bug: webrtc:9667 Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d Reviewed-on: https://webrtc-review.googlesource.com/96100 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24531}
2018-09-03 11:49:27 +02:00
"neteq/tools/neteq_test_factory.cc",
"neteq/tools/neteq_test_factory.h",
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
]
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 "Add AudioDecoderFactory to NetEqTest constructor." This is a reland of daa970f33e1923c5651a4a63c18e3d5361d0a795 Original change's description: > Add AudioDecoderFactory to NetEqTest constructor. > > Update EventLogAnalyzer to not depend on builtin audio decoders. > > Bug: webrtc:8396, webrtc:10080 > Change-Id: Ie02ed9cda6d4f11bfdf2e65eb6482283b7520738 > Reviewed-on: https://webrtc-review.googlesource.com/c/114301 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26026} Tbr: kwiberg@webrtc.org Bug: webrtc:8396, webrtc:10080 Change-Id: I598ce1cd41676b1992b0973b09476eeeb0e602d2 Reviewed-on: https://webrtc-review.googlesource.com/c/114940 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26058}
2018-12-19 15:06:17 +01:00
"../../api/audio_codecs:builtin_audio_decoder_factory",
Move NetEq headers to api/ This CL also introduces NetEqFactory and NetEqControllerFactory interfaces, as well as several convenience classes for working with them: DefaultNetEqFactory, DefaultNetEqControllerFactory and CustomNetEqFactory. Bug: webrtc:11005 Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29671}
2019-10-31 14:38:11 +01:00
"../../api/neteq:neteq_api",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Refactor NetEqTestFactory to not use "external" decoders Bug: webrtc:10080 Change-Id: Icfca98d6d91fc5139e678c1aa3de1e2c35abff5c Reviewed-on: https://webrtc-review.googlesource.com/c/115240 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26090}
2018-12-20 15:51:43 +01:00
"../../test:audio_codec_mocks",
Added support for skipping get_audio events, adding dummy packets and setting a field trial string. Bug: webrtc:10337 Change-Id: I0507da4d955daa914af774c946be16a4168be21a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150780 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29392}
2019-10-07 13:18:18 +02:00
"../../test:field_trial",
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
"../../test:test_support",
Restructure neteq_rtpplay into a library with small executable wrapper. Most of the code in neteq_rtpplay is moved into a factory class for NetEqTest. The factory method takes the same argc and argv arguments as neteq_rtpplay. This CL also adds a small public API for neteq_test to allow easy integration into external software. Bug: webrtc:9667 Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d Reviewed-on: https://webrtc-review.googlesource.com/96100 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24531}
2018-09-03 11:49:27 +02:00
]
}
Switch neteq_rtpplay into an executable. Bug: None Change-Id: Iecfe4f59336081583850ebcef9e115821a3f27f7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146715 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28677}
2019-07-25 09:48:14 +02:00
rtc_executable("neteq_rtpplay") {
Restructure neteq_rtpplay into a library with small executable wrapper. Most of the code in neteq_rtpplay is moved into a factory class for NetEqTest. The factory method takes the same argc and argv arguments as neteq_rtpplay. This CL also adds a small public API for neteq_test to allow easy integration into external software. Bug: webrtc:9667 Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d Reviewed-on: https://webrtc-review.googlesource.com/96100 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24531}
2018-09-03 11:49:27 +02:00
testonly = true
Mark neteq_rtpplay as publicly visible. Bug: None Change-Id: I051c7c23851ab15345c8e0f0322458d4f9a7e187 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128123 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27162}
2019-03-15 14:00:24 +01:00
visibility += [ "*" ]
Restructure neteq_rtpplay into a library with small executable wrapper. Most of the code in neteq_rtpplay is moved into a factory class for NetEqTest. The factory method takes the same argc and argv arguments as neteq_rtpplay. This CL also adds a small public API for neteq_test to allow easy integration into external software. Bug: webrtc:9667 Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d Reviewed-on: https://webrtc-review.googlesource.com/96100 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24531}
2018-09-03 11:49:27 +02:00
defines = []
deps = [
":neteq_test_factory",
":neteq_test_tools",
Move code for setting field trials from NetEqTestFactory to the main function in neteq_rtpplay. It is problematic to set field trials more than once, so to avoid running into problems, this functionality has been placed in the main function of neteq_rtpplay. Bug: webrtc:9667 Change-Id: Ib9b9990f30a1715b50889dbfc4d74787bcbe5dae Reviewed-on: https://webrtc-review.googlesource.com/98541 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24673}
2018-09-11 10:30:58 +02:00
"../../rtc_base:rtc_base_approved",
Reland "NetEQ RTP Play: Optionally write output audio file" This reverts commit c4b391a257ebf85448e58e73a96eb267635b6d6a. Reason for revert: issue fixed Original change's description: > Revert "NetEQ RTP Play: Optionally write output audio file" > > This reverts commit 6330818ec8159ee476481ba4a89f884fb3653f3f. > > Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots. > > Original change's description: > > NetEQ RTP Play: Optionally write output audio file > > > > This CL makes the output audio file optional to more > > quickly run neteq_rtpplay when no audio output is needed. > > The CL also includes necessary adaptations because of pre-existing > > dependencies (e.g., the output audio file name is used to create > > the plotting script file names). > > > > The command line arguments are retro-compatible - i.e., same behavior > > when specifying the output audio file and the new flag > > --output_files_base_name is not used. > > > > This CL also includes a test script with which the retro-compatibility > > has been verified. > > > > Bug: webrtc:10337 > > Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224 > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#27067} > > TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org > > Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10337 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293 > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27078} TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org Change-Id: Ia7061f7c2d69db61638ad612e82cd429eb49d539 Bug: webrtc:10337 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127540 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27106}
2019-03-13 11:51:44 +01:00
"../../rtc_base:stringutils",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:field_trial",
Move code for setting field trials from NetEqTestFactory to the main function in neteq_rtpplay. It is problematic to set field trials more than once, so to avoid running into problems, this functionality has been placed in the main function of neteq_rtpplay. Bug: webrtc:9667 Change-Id: Ib9b9990f30a1715b50889dbfc4d74787bcbe5dae Reviewed-on: https://webrtc-review.googlesource.com/98541 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24673}
2018-09-11 10:30:58 +02:00
"../../test:field_trial",
Switch neteq tools to ABSL_FLAG. Bug: webrtc:10616 Change-Id: I2aa688f0976d5618347e402f25d8701b0cf5a360 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144027 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28442}
2019-06-27 15:59:09 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
Reland "NetEQ RTP Play: Optionally write output audio file" This reverts commit c4b391a257ebf85448e58e73a96eb267635b6d6a. Reason for revert: issue fixed Original change's description: > Revert "NetEQ RTP Play: Optionally write output audio file" > > This reverts commit 6330818ec8159ee476481ba4a89f884fb3653f3f. > > Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots. > > Original change's description: > > NetEQ RTP Play: Optionally write output audio file > > > > This CL makes the output audio file optional to more > > quickly run neteq_rtpplay when no audio output is needed. > > The CL also includes necessary adaptations because of pre-existing > > dependencies (e.g., the output audio file name is used to create > > the plotting script file names). > > > > The command line arguments are retro-compatible - i.e., same behavior > > when specifying the output audio file and the new flag > > --output_files_base_name is not used. > > > > This CL also includes a test script with which the retro-compatibility > > has been verified. > > > > Bug: webrtc:10337 > > Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224 > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#27067} > > TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org > > Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10337 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293 > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27078} TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org Change-Id: Ia7061f7c2d69db61638ad612e82cd429eb49d539 Bug: webrtc:10337 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127540 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27106}
2019-03-13 11:51:44 +01:00
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
Restructure neteq_rtpplay into a library with small executable wrapper. Most of the code in neteq_rtpplay is moved into a factory class for NetEqTest. The factory method takes the same argc and argv arguments as neteq_rtpplay. This CL also adds a small public API for neteq_test to allow easy integration into external software. Bug: webrtc:9667 Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d Reviewed-on: https://webrtc-review.googlesource.com/96100 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24531}
2018-09-03 11:49:27 +02:00
]
sources = [
"neteq/tools/neteq_rtpplay.cc",
GN: Add neteq_rtpplay and rtc_event_log_source BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2081113003 Cr-Commit-Position: refs/heads/master@{#13239}
2016-06-21 05:38:42 -07:00
]
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
Adding missing resources to audio_codec_speed_tests. BUG=none Review-Url: https://codereview.webrtc.org/2727973004 Cr-Commit-Position: refs/heads/master@{#19168}
2017-07-27 05:49:57 -07:00
audio_codec_speed_tests_resources = [
"//resources/audio_coding/music_stereo_48kHz.pcm",
"//resources/audio_coding/speech_mono_16kHz.pcm",
"//resources/audio_coding/speech_mono_32_48kHz.pcm",
]
if (is_ios) {
bundle_data("audio_codec_speed_tests_data") {
testonly = true
sources = audio_codec_speed_tests_resources
outputs = [
"{{bundle_resources_dir}}/{{source_file_part}}",
]
}
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_test("audio_codec_speed_tests") {
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
testonly = true
defines = []
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
deps = [
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
]
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
sources = [
"codecs/isac/fix/test/isac_speed_test.cc",
"codecs/opus/opus_speed_test.cc",
"codecs/tools/audio_codec_speed_test.cc",
"codecs/tools/audio_codec_speed_test.h",
]
Adding missing resources to audio_codec_speed_tests. BUG=none Review-Url: https://codereview.webrtc.org/2727973004 Cr-Commit-Position: refs/heads/master@{#19168}
2017-07-27 05:49:57 -07:00
data = audio_codec_speed_tests_resources
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
GN: Update tests to have the correct shard timeout value on Android. TBR=mflodman@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2231413002 Cr-Commit-Position: refs/heads/master@{#13741}
2016-08-15 02:29:11 -07:00
shard_timeout = 900
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
}
Adding missing resources to audio_codec_speed_tests. BUG=none Review-Url: https://codereview.webrtc.org/2727973004 Cr-Commit-Position: refs/heads/master@{#19168}
2017-07-27 05:49:57 -07:00
if (is_ios) {
deps += [ ":audio_codec_speed_tests_data" ]
}
Migrated audio_codec_speed_tests target to GN. Review-Url: https://codereview.webrtc.org/2177373004 Cr-Commit-Position: refs/heads/master@{#13549}
2016-07-28 02:34:30 -07:00
deps += [
":isac_fix",
":webrtc_opus",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
Harmonize paths to dependent targets. This CL consistently use: * relative paths for WebRTC dependent targets (test_support) * absolute paths for shared dependent targets (abseil) This is a necessary (but insufficient) step to build WebRTC tests from Chromium tree (rtc_include_tests=true), since test/ doesn't sit anymore in the top level directory. We also make sure that target declarations and uses are consistent in regard to build_with_chromium flag. Bug: webrtc:9943 Bug: webrtc:9855 Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b Reviewed-on: https://webrtc-review.googlesource.com/c/108720 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25445}
2018-10-30 21:12:42 +01:00
"../../test:test_support",
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
"../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",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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_performance_test.cc",
"neteq/tools/neteq_performance_test.h",
]
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",
Move NetEq headers to api/ This CL also introduces NetEqFactory and NetEqControllerFactory interfaces, as well as several convenience classes for working with them: DefaultNetEqFactory, DefaultNetEqControllerFactory and CustomNetEqFactory. Bug: webrtc:11005 Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29671}
2019-10-31 14:38:11 +01:00
"../../api:neteq_factory_with_codecs",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_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",
Move NetEq headers to api/ This CL also introduces NetEqFactory and NetEqControllerFactory interfaces, as well as several convenience classes for working with them: DefaultNetEqFactory, DefaultNetEqControllerFactory and CustomNetEqFactory. Bug: webrtc:11005 Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29671}
2019-10-31 14:38:11 +01:00
"../../api/neteq:neteq_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../system_wrappers",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../test:test_support",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"//testing/gtest",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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",
]
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",
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",
Move NetEq headers to api/ This CL also introduces NetEqFactory and NetEqControllerFactory interfaces, as well as several convenience classes for working with them: DefaultNetEqFactory, DefaultNetEqControllerFactory and CustomNetEqFactory. Bug: webrtc:11005 Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29671}
2019-10-31 14:38:11 +01:00
"../../api/neteq:custom_neteq_factory",
"../../api/neteq:default_neteq_controller_factory",
"../../api/neteq:neteq_api",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" This reverts commit fab3460a821abe336ab610c6d6dfc0d392dac263. Reason for revert: fix downstream instead Original change's description: > Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" > > This reverts commit 9973933d2e606d64fcdc753acb9ba3afd6e30569. > > Reason for revert: breaking downstream projects and not reviewed by direct owners > > Original change's description: > > Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > This reverts commit 24192c267a40eb7d6b1850489ccdbf7a84f8ff0f. > > > > Reason for revert: Analyzed the performance regression in more detail. > > > > Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit 1796a820f60cb9429bf4bcf13a40a41794ac8fb0 has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac. > > > > There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall. > > > > Original change's description: > > > Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > > > This reverts commit 3e8ef940fe86cf6285afb80e68d2a0bedc631b9f. > > > > > > Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260. > > > > > > Original change's description: > > > > Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. > > > > > > > > This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time. > > > > > > > > Bug: webrtc:10668 > > > > Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890 > > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > > Reviewed-by: Minyue Li <minyue@webrtc.org> > > > > Commit-Queue: Chen Xing <chxg@google.com> > > > > Cr-Commit-Position: refs/heads/master@{#28434} > > > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > > > Bug: webrtc:10668, chromium:982260 > > > Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339 > > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28561} > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:10668, chromium:982260 > > Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707 > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Commit-Queue: Chen Xing <chxg@google.com> > > Cr-Commit-Position: refs/heads/master@{#28664} > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10668, chromium:982260 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28671} TBR=alessiob@webrtc.org,kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com Change-Id: Id43b7b3da79b4f48004b41767482bae1c1fa1e16 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10668, chromium:982260 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146713 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28672}
2019-07-24 16:47:02 +00:00
"../../system_wrappers",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../../test:test_support",
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",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
Refactor neteq_test_support. Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support. BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2252413002 Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 07:02:24 -07:00
]
}
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
rtc_executable("rtp_encode") {
Migrated target RTPencode to GN TBR=henrik.lundin@webrtc.org BUG=6162 Review-Url: https://codereview.webrtc.org/2195353002 Cr-Commit-Position: refs/heads/master@{#13599}
2016-08-02 01:45:50 -07:00
testonly = true
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
deps = audio_coding_deps + [
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
"//third_party/abseil-cpp/absl/memory",
Switch neteq tools to ABSL_FLAG. Bug: webrtc:10616 Change-Id: I2aa688f0976d5618347e402f25d8701b0cf5a360 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144027 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28442}
2019-06-27 15:59:09 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
":audio_coding",
Hide the AudioEncoderCng class behind a create function And put codecs/cng/webrtc_cng.h in a non-public build target while we're at it. Bug: webrtc:8396 Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c Reviewed-on: https://webrtc-review.googlesource.com/c/108782 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25486}
2018-11-01 11:13:44 +01:00
":audio_encoder_cng",
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
":neteq_input_audio_tools",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
"../../api/audio_codecs/g711:audio_encoder_g711",
"../../api/audio_codecs/L16:audio_encoder_L16",
"../../api/audio_codecs/g722:audio_encoder_g722",
"../../api/audio_codecs/ilbc:audio_encoder_ilbc",
"../../api/audio_codecs/isac:audio_encoder_isac",
"../../api/audio_codecs/opus:audio_encoder_opus",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"../../rtc_base:safe_conversions",
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
]
Migrated target RTPencode to GN TBR=henrik.lundin@webrtc.org BUG=6162 Review-Url: https://codereview.webrtc.org/2195353002 Cr-Commit-Position: refs/heads/master@{#13599}
2016-08-02 01:45:50 -07:00
sources = [
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
"neteq/tools/rtp_encode.cc",
Migrated target RTPencode to GN TBR=henrik.lundin@webrtc.org BUG=6162 Review-Url: https://codereview.webrtc.org/2195353002 Cr-Commit-Position: refs/heads/master@{#13599}
2016-08-02 01:45:50 -07:00
]
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
2017-11-24 09:28:57 +01:00
defines = audio_coding_defines
Migrated target RTPencode to GN TBR=henrik.lundin@webrtc.org BUG=6162 Review-Url: https://codereview.webrtc.org/2195353002 Cr-Commit-Position: refs/heads/master@{#13599}
2016-08-02 01:45:50 -07:00
}
Replacing the legacy tool RTPjitter with a new rtp_jitter This new tool provides the similar functionality as the legacy tool, but is implemented using less legacy helpers. It also replaces RTPtimeshift and RTPchange. The most significant change versus the old RTPjitter tool is that the new tool takes the timing data in the form of integers in a text file (instead of the binary data file used by the old tool). This should make it easier to create custom timing files when needed. Bug: webrtc:2692 Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287 Reviewed-on: https://webrtc-review.googlesource.com/25700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:39:39 +01:00
rtc_executable("rtp_jitter") {
Migrated RTPchange Gyp target to GN. BUG=6162 Review-Url: https://codereview.webrtc.org/2194213002 Cr-Commit-Position: refs/heads/master@{#13601}
2016-08-02 02:05:03 -07:00
testonly = true
Replacing the legacy tool RTPjitter with a new rtp_jitter This new tool provides the similar functionality as the legacy tool, but is implemented using less legacy helpers. It also replaces RTPtimeshift and RTPchange. The most significant change versus the old RTPjitter tool is that the new tool takes the timing data in the form of integers in a text file (instead of the binary data file used by the old tool). This should make it easier to create custom timing files when needed. Bug: webrtc:2692 Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287 Reviewed-on: https://webrtc-review.googlesource.com/25700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:39:39 +01:00
deps = audio_coding_deps + [
"../rtp_rtcp:rtp_rtcp_format",
"../../api:array_view",
"../../rtc_base:rtc_base_approved",
]
Migrated RTPchange Gyp target to GN. BUG=6162 Review-Url: https://codereview.webrtc.org/2194213002 Cr-Commit-Position: refs/heads/master@{#13601}
2016-08-02 02:05:03 -07:00
sources = [
Replacing the legacy tool RTPjitter with a new rtp_jitter This new tool provides the similar functionality as the legacy tool, but is implemented using less legacy helpers. It also replaces RTPtimeshift and RTPchange. The most significant change versus the old RTPjitter tool is that the new tool takes the timing data in the form of integers in a text file (instead of the binary data file used by the old tool). This should make it easier to create custom timing files when needed. Bug: webrtc:2692 Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287 Reviewed-on: https://webrtc-review.googlesource.com/25700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:39:39 +01:00
"neteq/tools/rtp_jitter.cc",
Migrated RTPchange Gyp target to GN. BUG=6162 Review-Url: https://codereview.webrtc.org/2194213002 Cr-Commit-Position: refs/heads/master@{#13601}
2016-08-02 02:05:03 -07:00
]
Replacing the legacy tool RTPjitter with a new rtp_jitter This new tool provides the similar functionality as the legacy tool, but is implemented using less legacy helpers. It also replaces RTPtimeshift and RTPchange. The most significant change versus the old RTPjitter tool is that the new tool takes the timing data in the form of integers in a text file (instead of the binary data file used by the old tool). This should make it easier to create custom timing files when needed. Bug: webrtc:2692 Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287 Reviewed-on: https://webrtc-review.googlesource.com/25700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:39:39 +01:00
defines = audio_coding_defines
Migrated RTPchange Gyp target to GN. BUG=6162 Review-Url: https://codereview.webrtc.org/2194213002 Cr-Commit-Position: refs/heads/master@{#13601}
2016-08-02 02:05:03 -07:00
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("rtpcat") {
Migrated GN target rtpcat. BUG=6162 Review-Url: https://codereview.webrtc.org/2202193002 Cr-Commit-Position: refs/heads/master@{#13605}
2016-08-02 04:27:25 -07:00
testonly = true
sources = [
"neteq/tools/rtpcat.cc",
]
deps = [
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Migrated GN target rtpcat. BUG=6162 Review-Url: https://codereview.webrtc.org/2202193002 Cr-Commit-Position: refs/heads/master@{#13605}
2016-08-02 04:27:25 -07:00
"../../test:rtp_test_utils",
"//testing/gtest",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("rtp_analyze") {
GN migration of target audio_coding/neteq/rtp_analyze. Review-Url: https://codereview.webrtc.org/2185533003 Cr-Commit-Position: refs/heads/master@{#13533}
2016-07-26 06:46:19 -07:00
testonly = true
sources = [
"neteq/tools/rtp_analyze.cc",
]
deps = [
":neteq",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_test_tools",
GN migration of target audio_coding/neteq/rtp_analyze. Review-Url: https://codereview.webrtc.org/2185533003 Cr-Commit-Position: refs/heads/master@{#13533}
2016-07-26 06:46:19 -07:00
":pcm16b",
"//testing/gtest",
Switch neteq tools to ABSL_FLAG. Bug: webrtc:10616 Change-Id: I2aa688f0976d5618347e402f25d8701b0cf5a360 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144027 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28442}
2019-06-27 15:59:09 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
GN migration of target audio_coding/neteq/rtp_analyze. Review-Url: https://codereview.webrtc.org/2185533003 Cr-Commit-Position: refs/heads/master@{#13533}
2016-07-26 06:46:19 -07:00
]
}
Migrated GN target :neteq_opus_quality_test Migrated GN target :neteq_opus_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190, webrtc:2692 Review-Url: https://codereview.webrtc.org/2223933004 Cr-Commit-Position: refs/heads/master@{#13701}
2016-08-10 02:11:30 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("neteq_opus_quality_test") {
Migrated GN target :neteq_opus_quality_test Migrated GN target :neteq_opus_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190, webrtc:2692 Review-Url: https://codereview.webrtc.org/2223933004 Cr-Commit-Position: refs/heads/master@{#13701}
2016-08-10 02:11:30 -07:00
testonly = true
sources = [
"neteq/test/neteq_opus_quality_test.cc",
]
deps = [
":neteq",
Refactor neteq_test_support. Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support. BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2252413002 Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 07:02:24 -07:00
":neteq_quality_test_support",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_tools",
Migrated GN target :neteq_opus_quality_test Migrated GN target :neteq_opus_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190, webrtc:2692 Review-Url: https://codereview.webrtc.org/2223933004 Cr-Commit-Position: refs/heads/master@{#13701}
2016-08-10 02:11:30 -07:00
":webrtc_opus",
Replace gflags usages with rtc_base/flags in all targets based on test_main BUG=webrtc:7644 Review-Url: https://codereview.webrtc.org/2995363002 Cr-Commit-Position: refs/heads/master@{#19580}
2017-08-29 05:51:57 -07:00
"../../rtc_base:rtc_base_approved",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
Migrated GN target :neteq_opus_quality_test Migrated GN target :neteq_opus_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190, webrtc:2692 Review-Url: https://codereview.webrtc.org/2223933004 Cr-Commit-Position: refs/heads/master@{#13701}
2016-08-10 02:11:30 -07:00
"//testing/gtest",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
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
]
}
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",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"../../rtc_base:checks",
Migrated GN target :neteq_speed_test Migrated GN target :neteq_speed_test from webrtc/modules/audio_coding/neteq/neteq.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2223343004 Cr-Commit-Position: refs/heads/master@{#13704}
2016-08-10 04:16:36 -07:00
"../../test:test_support",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
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",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
Migrated GN target :neteq_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",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
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
]
}
Migrated GN target :neteq_isac_quality_test Migrated GN target :neteq_isac_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2226253002 Cr-Commit-Position: refs/heads/master@{#13710}
2016-08-10 05:04:47 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("neteq_isac_quality_test") {
Migrated GN target :neteq_isac_quality_test Migrated GN target :neteq_isac_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2226253002 Cr-Commit-Position: refs/heads/master@{#13710}
2016-08-10 05:04:47 -07:00
testonly = true
sources = [
"neteq/test/neteq_isac_quality_test.cc",
]
deps = [
":isac_fix",
":neteq",
Refactor neteq_test_support. Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support. BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2252413002 Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 07:02:24 -07:00
":neteq_quality_test_support",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
Migrated GN target :neteq_isac_quality_test Migrated GN target :neteq_isac_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2226253002 Cr-Commit-Position: refs/heads/master@{#13710}
2016-08-10 05:04:47 -07:00
"//testing/gtest",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
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
]
}
Migrated GN target :neteq_pcmu_quality_test Migrated GN target :neteq_pcmu_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2224313002 Cr-Commit-Position: refs/heads/master@{#13711}
2016-08-10 05:06:27 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("neteq_pcmu_quality_test") {
Migrated GN target :neteq_pcmu_quality_test Migrated GN target :neteq_pcmu_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2224313002 Cr-Commit-Position: refs/heads/master@{#13711}
2016-08-10 05:06:27 -07:00
testonly = true
sources = [
"neteq/test/neteq_pcmu_quality_test.cc",
]
deps = [
":g711",
":neteq",
Refactor neteq_test_support. Take 'tools/neteq_quality_test.cc' and 'tools/neteq_quality_test.h' outside of neteq_test_support into their own target, neteq_quality_test_support. BUG=webrtc:6228 NOTRY=True Review-Url: https://codereview.webrtc.org/2252413002 Cr-Commit-Position: refs/heads/master@{#13831}
2016-08-19 07:02:24 -07:00
":neteq_quality_test_support",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
Migrated GN target :neteq_pcmu_quality_test Migrated GN target :neteq_pcmu_quality_test from webrtc/modules/audio_coding/neteq/neteq_tests.gypi NOTRY=True BUG=webrtc:6190 Review-Url: https://codereview.webrtc.org/2224313002 Cr-Commit-Position: refs/heads/master@{#13711}
2016-08-10 05:06:27 -07:00
"//testing/gtest",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
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
]
}
Migrated GN target :isac_fix_test Migrated GN target :isac_fix_test from webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2247233002 Cr-Commit-Position: refs/heads/master@{#13770}
2016-08-16 01:47:16 -07:00
Add pcm16b quality test supporting 48khz. Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Bug: webrtc:8836 Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Reviewed-on: https://webrtc-review.googlesource.com/47400 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21878}
2018-02-02 17:21:46 +01:00
rtc_executable("neteq_pcm16b_quality_test") {
testonly = true
sources = [
"neteq/test/neteq_pcm16b_quality_test.cc",
]
deps = [
":neteq",
":neteq_quality_test_support",
":pcm16b",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Add pcm16b quality test supporting 48khz. Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Bug: webrtc:8836 Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Reviewed-on: https://webrtc-review.googlesource.com/47400 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21878}
2018-02-02 17:21:46 +01:00
"../../test:test_main",
"//testing/gtest",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
Add pcm16b quality test supporting 48khz. Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Bug: webrtc:8836 Change-Id: I7abb394c61f6fd260f060ca5c16167ae6b44ef68 Reviewed-on: https://webrtc-review.googlesource.com/47400 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21878}
2018-02-02 17:21:46 +01:00
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("isac_fix_test") {
Migrated GN target :isac_fix_test Migrated GN target :isac_fix_test from webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2247233002 Cr-Commit-Position: refs/heads/master@{#13770}
2016-08-16 01:47:16 -07:00
testonly = true
sources = [
"codecs/isac/fix/test/kenny.cc",
]
deps = [
":isac_fix",
Reland "iOS: Save perf results under Documents/perf_result.json" This will require a manual roll to downstream projects, since the //test:perf_test target was introduced. This is a reland of 10a8e7a9b5261a7e3ce19900ba3511be3b5911f8 Original change's description: > iOS: Save perf results under Documents/perf_result.json > > TBR=henrika@webrtc.org > > Bug: webrtc:7156 > Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5 > Reviewed-on: https://webrtc-review.googlesource.com/29202 > Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21244} R=henrika@webrtc.org, phoglund@webrtc.org Bug: webrtc:7156 Change-Id: I85fc7bc5fce0894af90017b71b9952b61b523424 Reviewed-on: https://webrtc-review.googlesource.com/37643 Reviewed-by: Patrik Höglund <phoglund@google.com> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21518}
2018-01-05 15:34:09 +01:00
"../../test:perf_test",
Migrated GN target :isac_fix_test Migrated GN target :isac_fix_test from webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2247233002 Cr-Commit-Position: refs/heads/master@{#13770}
2016-08-16 01:47:16 -07:00
"../../test:test_support",
]
Make it possible to isolate isac_fix_test. Successful swarming run: https://chromium-swarm.appspot.com/task?id=38769e8dd5d3e110&refresh=10&show_raw=1 BUG=chromium:755660 NOTRY=True Review-Url: https://codereview.webrtc.org/3009203002 Cr-Commit-Position: refs/heads/master@{#19738}
2017-09-07 23:18:35 -07:00
data = [
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
"../../resources/speech_and_misc_wb.pcm",
Make it possible to isolate isac_fix_test. Successful swarming run: https://chromium-swarm.appspot.com/task?id=38769e8dd5d3e110&refresh=10&show_raw=1 BUG=chromium:755660 NOTRY=True Review-Url: https://codereview.webrtc.org/3009203002 Cr-Commit-Position: refs/heads/master@{#19738}
2017-09-07 23:18:35 -07:00
]
Migrated GN target :isac_fix_test Migrated GN target :isac_fix_test from webrtc/modules/audio_coding/codecs/isac/isacfix_test.gypi NOTRY=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2247233002 Cr-Commit-Position: refs/heads/master@{#13770}
2016-08-16 01:47:16 -07:00
}
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
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("isac_test_util") {
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
testonly = true
sources = [
"codecs/isac/main/util/utility.c",
Add missing iSAC headers. Bug: webrtc:7619 Change-Id: I08df7774ca7e333e84bb5ca97805181f375af942 Reviewed-on: https://webrtc-review.googlesource.com/34647 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21467}
2018-01-02 11:49:48 +01:00
"codecs/isac/main/util/utility.h",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("isac_test") {
Added GN target for isac_test. BUG=webrtc:6191 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2267423002 Cr-Commit-Position: refs/heads/master@{#13884}
2016-08-24 02:25:57 -07:00
testonly = true
sources = [
"codecs/isac/main/test/simpleKenny.c",
]
deps = [
":isac",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
":isac_test_util",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Added GN target for isac_test. BUG=webrtc:6191 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2267423002 Cr-Commit-Position: refs/heads/master@{#13884}
2016-08-24 02:25:57 -07:00
]
}
GN: 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",
]
}
Added GN target for webrtc_opus_fec_test. BUG=webrtc:6191 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2268213002 Cr-Commit-Position: refs/heads/master@{#13893}
2016-08-24 06:11:18 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("isac_api_test") {
Migrated ILBC and ISAC test targets for GN. Migrated GN targets ilbc_test, isac_api_test, isac_switch_samprate_test from webrtc/modules/audio_coding/codecs NOTRY=True NOPRESUBMIT=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2270403002 Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 04:09:19 -07:00
testonly = true
sources = [
"codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc",
]
deps = [
":isac",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
":isac_test_util",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Migrated ILBC and ISAC test targets for GN. Migrated GN targets ilbc_test, isac_api_test, isac_switch_samprate_test from webrtc/modules/audio_coding/codecs NOTRY=True NOPRESUBMIT=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2270403002 Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 04:09:19 -07:00
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("isac_switch_samprate_test") {
Migrated ILBC and ISAC test targets for GN. Migrated GN targets ilbc_test, isac_api_test, isac_switch_samprate_test from webrtc/modules/audio_coding/codecs NOTRY=True NOPRESUBMIT=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2270403002 Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 04:09:19 -07:00
testonly = true
sources = [
"codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc",
]
deps = [
":isac",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
":isac_test_util",
Using fully qualified #include paths in isac code. WebRTC internal code should always used include paths that starts from the root of the project and that clearly identify the header file. This allows 'gn check' to actually keep dependencies under control because 'gn check' cannot enforce anything if the include path is not fully qualified (starting from the root of the project). Bug: webrtc:8815 Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc Reviewed-on: https://webrtc-review.googlesource.com/46101 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21847}
2018-02-01 14:12:55 +01:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Migrated ILBC and ISAC test targets for GN. Migrated GN targets ilbc_test, isac_api_test, isac_switch_samprate_test from webrtc/modules/audio_coding/codecs NOTRY=True NOPRESUBMIT=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2270403002 Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 04:09:19 -07:00
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("ilbc_test") {
Migrated ILBC and ISAC test targets for GN. Migrated GN targets ilbc_test, isac_api_test, isac_switch_samprate_test from webrtc/modules/audio_coding/codecs NOTRY=True NOPRESUBMIT=True BUG=webrtc:6191 Review-Url: https://codereview.webrtc.org/2270403002 Cr-Commit-Position: refs/heads/master@{#13953}
2016-08-29 04:09:19 -07:00
testonly = true
sources = [
"codecs/ilbc/test/iLBC_test.c",
]
deps = [
":ilbc",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("webrtc_opus_fec_test") {
Added GN target for webrtc_opus_fec_test. BUG=webrtc:6191 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2268213002 Cr-Commit-Position: refs/heads/master@{#13893}
2016-08-24 06:11:18 -07:00
testonly = true
sources = [
"codecs/opus/opus_fec_test.cc",
]
deps = [
":webrtc_opus",
"../../common_audio",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac test_support_main_threaded_mac doesn't seem to be used. It looks like it was last used about a year and a half ago, and was removed in https://webrtc-codereview.appspot.com/55379004 BUG=webrtc:6424 R=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2540693002 Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 06:12:01 -08:00
"../../test:test_main",
Harmonize paths to dependent targets. This CL consistently use: * relative paths for WebRTC dependent targets (test_support) * absolute paths for shared dependent targets (abseil) This is a necessary (but insufficient) step to build WebRTC tests from Chromium tree (rtc_include_tests=true), since test/ doesn't sit anymore in the top level directory. We also make sure that target declarations and uses are consistent in regard to build_with_chromium flag. Bug: webrtc:9943 Bug: webrtc:9855 Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b Reviewed-on: https://webrtc-review.googlesource.com/c/108720 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25445}
2018-10-30 21:12:42 +01:00
"../../test:test_support",
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: 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
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_coding_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
testonly = true
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility += webrtc_default_visibility
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
sources = [
"acm2/acm_receiver_unittest.cc",
"acm2/audio_coding_module_unittest.cc",
"acm2/call_statistics_unittest.cc",
"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",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"audio_network_adaptor/frame_length_controller_unittest.cc",
Avoid code duplication between PLR/RPLR-based FecController BUG=webrtc:7058 Review-Url: https://codereview.webrtc.org/2688613003 Cr-Commit-Position: refs/heads/master@{#17437}
2017-03-29 03:16:58 -07:00
"audio_network_adaptor/util/threshold_curve_unittest.cc",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"codecs/builtin_audio_decoder_factory_unittest.cc",
Injectable audio encoders: BuiltinAudioEncoderFactory This CL contains all the changes made to audio_coding while making audio encoders injectable. Apart from some small changes to webrtcvoiceengine, nothing here is hooked up to the outside world. Those changes will be added to a follow-up CL. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2695243005 Cr-Commit-Position: refs/heads/master@{#17569}
2017-04-06 10:03:21 -07:00
"codecs/builtin_audio_encoder_factory_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"codecs/cng/audio_encoder_cng_unittest.cc",
"codecs/cng/cng_unittest.cc",
"codecs/ilbc/ilbc_unittest.cc",
"codecs/isac/fix/source/filterbanks_unittest.cc",
"codecs/isac/fix/source/filters_unittest.cc",
"codecs/isac/fix/source/lpc_masking_model_unittest.cc",
"codecs/isac/fix/source/transform_unittest.cc",
"codecs/isac/main/source/audio_encoder_isac_unittest.cc",
"codecs/isac/main/source/isac_unittest.cc",
"codecs/legacy_encoded_audio_frame_unittest.cc",
Decoder for multistream Opus. See https://webrtc-review.googlesource.com/c/src/+/121764 for the overall vision. This CL adds a multistream Opus decoder. It's a new code-path to not interfere with the standard Opus decoder. We introduce new SDP syntax, which uses terminology of RFC 7845. We also set up the decoder side to parse it. The encoder part will come in a later CL. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 17:19:41 +02:00
"codecs/opus/audio_decoder_multi_channel_opus_unittest.cc",
Encoder side of Multistream Opus. Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely. Adds an ENCODER and sets it up to parse SDP config for multistream opus. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27775}
2019-04-25 15:09:32 +02:00
"codecs/opus/audio_encoder_multi_channel_opus_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"codecs/opus/audio_encoder_opus_unittest.cc",
Implement Opus bandwidth adjustment behind a FieldTrial Bug: webrtc:8522 Change-Id: I3a32ebfecd27ff74b507c2cee9e16aab17153442 Reviewed-on: https://webrtc-review.googlesource.com/22210 Commit-Queue: Alejandro Luebs <aluebs@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20799}
2017-11-20 11:13:56 -08:00
"codecs/opus/opus_bandwidth_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"codecs/opus/opus_unittest.cc",
"codecs/red/audio_encoder_copy_red_unittest.cc",
"neteq/audio_multi_vector_unittest.cc",
"neteq/audio_vector_unittest.cc",
"neteq/background_noise_unittest.cc",
"neteq/buffer_level_filter_unittest.cc",
"neteq/comfort_noise_unittest.cc",
"neteq/decision_logic_unittest.cc",
"neteq/decoder_database_unittest.cc",
"neteq/delay_manager_unittest.cc",
"neteq/delay_peak_detector_unittest.cc",
"neteq/dsp_helper_unittest.cc",
"neteq/dtmf_buffer_unittest.cc",
"neteq/dtmf_tone_generator_unittest.cc",
"neteq/expand_unittest.cc",
Refactor DelayManager into separate Histogram class and make it injectable for testing purposes. Change-Id: I98aa3f992169e598fc1a3dd850400183395fe1fe Bug: webrtc:10333 Reviewed-on: https://webrtc-review.googlesource.com/c/123445 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26797}
2019-02-21 15:42:31 +01:00
"neteq/histogram_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"neteq/merge_unittest.cc",
"neteq/mock/mock_decoder_database.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",
Refactor DelayManager into separate Histogram class and make it injectable for testing purposes. Change-Id: I98aa3f992169e598fc1a3dd850400183395fe1fe Bug: webrtc:10333 Reviewed-on: https://webrtc-review.googlesource.com/c/123445 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26797}
2019-02-21 15:42:31 +01:00
"neteq/mock/mock_histogram.h",
Introduce injectable NetEqController interface. This interface is implemented by the DecisionLogic class, which now contains the DelayManager and DelayPeakDetector. Bug: webrtc:11005 Change-Id: I4fb69fa359e60831cf153e41f101d5b623749380 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155176 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29613}
2019-10-24 15:20:39 +02:00
"neteq/mock/mock_neteq_controller.h",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"neteq/mock/mock_packet_buffer.h",
"neteq/mock/mock_red_payload_splitter.h",
Implement packet discard rate in NetEq. BUG=webrtc:7903 Change-Id: I819c9362671ca0b02c602d53e4dc39afdd8ec465 Reviewed-on: https://chromium-review.googlesource.com/555311 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18899}
2017-07-05 11:17:40 +02:00
"neteq/mock/mock_statistics_calculator.h",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"neteq/nack_tracker_unittest.cc",
Let NetEq use the PLC output from a decoder This change enables NetEq to use the packet concealment audio (aka PLC) produced by a decoder. The change also includes a new API to the AudioDecoder interface, which lets the decoder implementation generate and deliver concealment audio. Bug: webrtc:9180 Change-Id: Icaacebccf645d4694b0d2d6310f6f2c7132881c4 Reviewed-on: https://webrtc-review.googlesource.com/96340 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24738}
2018-09-05 18:14:52 +02:00
"neteq/neteq_decoder_plc_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"neteq/neteq_impl_unittest.cc",
"neteq/neteq_network_stats_unittest.cc",
"neteq/neteq_stereo_unittest.cc",
"neteq/neteq_unittest.cc",
"neteq/normal_unittest.cc",
"neteq/packet_buffer_unittest.cc",
"neteq/post_decode_vad_unittest.cc",
"neteq/random_vector_unittest.cc",
"neteq/red_payload_splitter_unittest.cc",
NetEq/Stats: Don't let concealed_samples decrease When NetEq performs a merge operation, it will usually have to correct the stats for number of concealment samples produced, sometimes with decreasing it. This does not make sense in the context of the stats spec, and stats-consuming applications may not be prepared for it. With this change, only positive corrections are allowed for the concealed_samples value. This will sometimes lead to a small positive bias, but it will be negligible over time. Bug: webrtc:8253 Change-Id: Ie9de311ab16401f1a4b435f6269725901b8cf561 Reviewed-on: https://webrtc-review.googlesource.com/1583 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19941}
2017-09-25 12:22:46 +02:00
"neteq/statistics_calculator_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"neteq/sync_buffer_unittest.cc",
"neteq/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",
Decoder for multistream Opus. See https://webrtc-review.googlesource.com/c/src/+/121764 for the overall vision. This CL adds a multistream Opus decoder. It's a new code-path to not interfere with the standard Opus decoder. We introduce new SDP syntax, which uses terminology of RFC 7845. We also set up the decoder side to parse it. The encoder part will come in a later CL. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 17:19:41 +02:00
":audio_coding_opus_common",
Hide the AudioEncoderCng class behind a create function And put codecs/cng/webrtc_cng.h in a non-public build target while we're at it. Bug: webrtc:8396 Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c Reviewed-on: https://webrtc-review.googlesource.com/c/108782 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25486}
2018-11-01 11:13:44 +01:00
":audio_encoder_cng",
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",
":g711",
":ilbc",
iSAC: Untangle some cyclic dependencies BUG=webrtc:6828, webrtc:7042 Review-Url: https://codereview.webrtc.org/2656823005 Cr-Commit-Position: refs/heads/master@{#16358}
2017-01-30 05:28:54 -08:00
":isac",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":isac_c",
":isac_fix",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
":legacy_encoded_audio_frame",
Add missing files to audio_coding Bug: webrtc:7650 Change-Id: I8d7d8c3998799404ec6283896883d195468cdfdc Reviewed-on: https://webrtc-review.googlesource.com/37622 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21502}
2018-01-05 11:46:26 +01:00
":mocks",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":neteq",
":neteq_test_support",
NetEq tests: BUILD target reorg In this CL, the neteq_unittest_tools target is split in two separate targets. One still called neteq_tools which does not set testonly=true and that includes code related to audio input, replacement audio and fake decoding. The other target called neteq_test_tools contains the remaining files, and is still under testonly=true. Other renames: neteq_test_tools -> neteq_test_tools_deprecated neteq_test_minimal -> neteq_tools_minimal Cyclic dependencies were also cleaned up. CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng BUG=webrtc:7467,webrtc:6828 Review-Url: https://codereview.webrtc.org/2845013003 Cr-Commit-Position: refs/heads/master@{#17921}
2017-04-28 00:59:45 -07:00
":neteq_test_tools",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":pcm16b",
":red",
Hide the AudioEncoderCng class behind a create function And put codecs/cng/webrtc_cng.h in a non-public build target while we're at it. Bug: webrtc:8396 Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c Reviewed-on: https://webrtc-review.googlesource.com/c/108782 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25486}
2018-11-01 11:13:44 +01:00
":webrtc_cng",
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_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",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"..:module_api_public",
Move NetEq headers to api/ This CL also introduces NetEqFactory and NetEqControllerFactory interfaces, as well as several convenience classes for working with them: DefaultNetEqFactory, DefaultNetEqControllerFactory and CustomNetEqFactory. Bug: webrtc:11005 Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29671}
2019-10-31 14:38:11 +01:00
"../../api:neteq_factory_with_codecs",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_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",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../../api/audio_codecs:builtin_audio_encoder_factory",
Decoder for multistream Opus. See https://webrtc-review.googlesource.com/c/src/+/121764 for the overall vision. This CL adds a multistream Opus decoder. It's a new code-path to not interfere with the standard Opus decoder. We introduce new SDP syntax, which uses terminology of RFC 7845. We also set up the decoder side to parse it. The encoder part will come in a later CL. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27493}
2019-04-08 17:19:41 +02:00
"../../api/audio_codecs/opus:audio_decoder_multiopus",
Implement Opus bandwidth adjustment behind a FieldTrial Bug: webrtc:8522 Change-Id: I3a32ebfecd27ff74b507c2cee9e16aab17153442 Reviewed-on: https://webrtc-review.googlesource.com/22210 Commit-Queue: Alejandro Luebs <aluebs@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20799}
2017-11-20 11:13:56 -08:00
"../../api/audio_codecs/opus:audio_decoder_opus",
Encoder side of Multistream Opus. Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely. Adds an ENCODER and sets it up to parse SDP config for multistream opus. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27775}
2019-04-25 15:09:32 +02:00
"../../api/audio_codecs/opus:audio_encoder_multiopus",
Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix We've done this previously with the other audio encoders, but Opus had to wait until all external users had been updated. BUG=webrtc:7847 Change-Id: I70422d7b6c715f32a43bee88febcf6b6155e18b3 Reviewed-on: https://webrtc-review.googlesource.com/8000 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20424}
2017-10-25 09:57:40 +02:00
"../../api/audio_codecs/opus:audio_encoder_opus",
Move NetEq headers to api/ This CL also introduces NetEqFactory and NetEqControllerFactory interfaces, as well as several convenience classes for working with them: DefaultNetEqFactory, DefaultNetEqControllerFactory and CustomNetEqFactory. Bug: webrtc:11005 Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29671}
2019-10-31 14:38:11 +01:00
"../../api/neteq:custom_neteq_factory",
"../../api/neteq:default_neteq_controller_factory",
"../../api/neteq:neteq_api",
"../../api/neteq:neteq_controller_api",
"../../api/neteq:tick_timer",
"../../api/neteq:tick_timer_unittest",
Stop using rtc_event.h forward header Bug: webrtc:10206 Change-Id: I16905ec745673178195d6715fda6175c31500163 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151601 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29149}
2019-09-05 11:29:59 +02:00
"../../api/rtc_event_log",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Fix Gn Untracked headers in webrtc/common_audio Fixed following headers in this CL =================================== src/webrtc/common_audio/vad/mock/mock_vad.h src/webrtc/common_audio/mocks/mock_smoothing_filter.h src/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h BUG=webrtc:7648 Review-Url: https://codereview.webrtc.org/3013063002 Cr-Original-Commit-Position: refs/heads/master@{#19824} Review-Url: https://codereview.webrtc.org/3013673002 Cr-Commit-Position: refs/heads/master@{#19852}
2017-09-15 03:51:34 -07:00
"../../common_audio:mock_common_audio",
Reland: Add mock_rtc_event_log.h. Bug: webrtc:7642 Change-Id: I3f97a8b603e34819e1563b335c1eeb51f89b11ac Reviewed-on: https://webrtc-review.googlesource.com/37081 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21480}
2018-01-03 09:08:20 +01:00
"../../logging:mocks",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../../logging:rtc_event_audio",
Move RtcpStatistics from common_types.h to a new header file New location is modules/rtp_rtcp/include/rtcp_statistics.h. Bug: webrtc:5876 Change-Id: I85f55b58658588228ed77175226b3479352fd5de Reviewed-on: https://webrtc-review.googlesource.com/c/111961 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25799}
2018-11-27 14:05:08 +01:00
"../../modules/rtp_rtcp:rtp_rtcp_format",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../rtc_base",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Move rtc_base/ignore_wundef.h to its own target. Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to move it to its own target and allow its dependant to avoid to take a dependency rtc_base:on rtc_base_approved. Bug: webrtc:9419 Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29548}
2019-10-19 12:03:23 +02:00
"../../rtc_base:ignore_wundef",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_base_tests_utils",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:sanitizer",
Add some missing includes and dependencies. In particular, time_utils.h is currently pulled in via rtc_event.h This CL is in preparation of moving parts of the RTC event log to api/. Bug: webrtc:10206 Change-Id: Idd35aa9404afded4d29b1296344996c45b8c2e91 Reviewed-on: https://webrtc-review.googlesource.com/c/117921 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26326}
2019-01-16 17:45:05 +01:00
"../../rtc_base:timeutils",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
"../../system_wrappers:cpu_features_api",
Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ AudioDecoder and AudioDecoderFactory are in webrtc/api/ now, so move their mocks to someplace central where tests from all over WebRTC are allowed to #include them. BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2798063004 Cr-Commit-Position: refs/heads/master@{#17619}
2017-04-10 05:15:48 -07:00
"../../test:audio_codec_mocks",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:field_trial",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:rtp_test_utils",
"../../test:test_common",
"../../test:test_support",
Moving LappedTransform, Blocker and AudioRingBuffer. LappedTransform is only used in BandwidthAdaptationTest and therefore it should not be anymore a visible target under common_audio. This CL moves LappedTransform and other two classes it depends on (and which are not used elsewhere) to modules/audio_coding/codecs/opus/test. Bug: webrtc:9577, webrtc:5298 Change-Id: I1aa8052c2df2b2b150c279c0c9b1001474aed47a Reviewed-on: https://webrtc-review.googlesource.com/96440 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24509}
2018-08-31 10:41:37 +02:00
"codecs/opus/test",
"codecs/opus/test:test_unittest",
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
"//testing/gtest",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Depend directly on destination targets. Makes 'gn check' happy. Followup to https://webrtc-review.googlesource.com/c/src/+/106820 Bug: webrtc:5876, webrtc:9855 Change-Id: I33fa2c31ba26dc10c9a9c17da0ffed255c1f4d5b Reviewed-on: https://webrtc-review.googlesource.com/c/108760 Commit-Queue: Yves Gerey <yvesg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25447}
2018-10-30 22:23:02 +01:00
"//third_party/abseil-cpp/absl/types:optional",
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
]
defines = audio_coding_defines
if (rtc_enable_protobuf) {
Replacing WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP with WEBRTC_ENABLE_PROTOBUF. Bug: None Change-Id: I595b094e7fcb12723614df3197a40833932ba0a0 Reviewed-on: https://chromium-review.googlesource.com/533074 Reviewed-by: Michael T <tschumim@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18568}
2017-06-13 11:49:29 +02:00
defines += [ "WEBRTC_NETEQ_UNITTEST_BITEXACT" ]
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
deps += [
":ana_config_proto",
":neteq_unittest_proto",
]
}
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
# For backwards compatibility only! Use
# webrtc/api/audio_codecs:audio_codecs_api instead.
# TODO(kwiberg): Remove this.
rtc_source_set("audio_decoder_interface") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
sources = [
"codecs/audio_decoder.h",
]
deps = [
"../../api/audio_codecs:audio_codecs_api",
]
}
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
# For backwards compatibility only! Use
# webrtc/api/audio_codecs:audio_codecs_api instead.
# TODO(ossu): Remove this.
rtc_source_set("audio_encoder_interface") {
Restrict visibility in some places where we can get away with doing so BUG=webrtc:8255 Change-Id: I091a43703b7b7a75406ba58afb505f9b631a5521 Reviewed-on: https://webrtc-review.googlesource.com/10810 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21226}
2017-12-12 10:41:29 +01:00
visibility += [ "*" ]
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
sources = [
"codecs/audio_encoder.h",
]
deps = [
"../../api/audio_codecs:audio_codecs_api",
]
}
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