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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 += [ "*" ]
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
sources = [ "include/audio_coding_module_typedefs.h" ]
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",
ACM: Adding unittests for the remixing functionality On top of adding unittests for the remixing, the CL moves the code tested to a separate file in order to allow it to be tested. Bug: webrtc:11007 Change-Id: I531736517bbcc715b3c1bf3a4256c42208c5b778 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155740 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29839}
2019-11-19 21:00:59 +01:00
"acm2/acm_remixing.cc",
"acm2/acm_remixing.h",
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_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",
Add a CreateNetEq method that takes an AudioDecoderFactory The NetEqFactory is currently expected to wrap the AudioDecoderFactory, but this turns out not to be a good idea. Instead, it makes more sense to pass the AudioDecoderFactory through the CreateNetEq method. Bug: webrtc:11005 Change-Id: I8027ff6593f40c92072e7e88157631dcf329a984 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160644 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29918}
2019-11-26 12:29:05 +01:00
":default_neteq_factory",
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
":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: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",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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
"../../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I03b78bd2e411e9bcca199f85e4457511826cd17e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176745 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31649}
2020-07-07 15:53:34 +02:00
"../../rtc_base/synchronization:mutex",
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
"../../system_wrappers",
"../../system_wrappers:metrics",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
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
"//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",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
[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",
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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
[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",
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",
Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. The WebRTC-SendSideBwe-WithOverhead field trial requires audio encoders to properly implement the AudioEncoder::GetFrameLengthRange() function. Thic CL implements the function for all audio encoders in WebRTC in preparation for making that function pure virtual in the interface. Bug: webrtc:11427 Change-Id: Ieab6b6c72c62af6ac9525a20fcb39bd477079551 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171503 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30890}
2020-03-24 16:00:51 +01:00
"../../api/units:time_delta",
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
"../../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 absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//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
}
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",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../../api:field_trials_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",
Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. The WebRTC-SendSideBwe-WithOverhead field trial requires audio encoders to properly implement the AudioEncoder::GetFrameLengthRange() function. Thic CL implements the function for all audio encoders in WebRTC in preparation for making that function pure virtual in the interface. Bug: webrtc:11427 Change-Id: Ieab6b6c72c62af6ac9525a20fcb39bd477079551 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171503 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30890}
2020-03-24 16:00:51 +01:00
"../../api/units:time_delta",
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",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
Move byte_order out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ieb2fbaad8831439ca04fc5249e295c8839f6890e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259309 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36633}
2022-04-19 17:18:36 +02:00
"../../rtc_base:byte_order",
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 //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
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
]
Adopt absl::string_view in modules/audio_coding/ Bug: webrtc:13579 Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37573}
2022-07-20 12:53:07 +02:00
absl_deps = [
"//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("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",
Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. The WebRTC-SendSideBwe-WithOverhead field trial requires audio encoders to properly implement the AudioEncoder::GetFrameLengthRange() function. Thic CL implements the function for all audio encoders in WebRTC in preparation for making that function pure virtual in the interface. Bug: webrtc:11427 Change-Id: Ieab6b6c72c62af6ac9525a20fcb39bd477079551 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171503 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30890}
2020-03-24 16:00:51 +01:00
"../../api/units:time_delta",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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",
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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
public_deps = [ ":g711_c" ] # no-presubmit-check TODO(webrtc:8603)
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
}
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",
]
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
deps = [ "../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",
Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. The WebRTC-SendSideBwe-WithOverhead field trial requires audio encoders to properly implement the AudioEncoder::GetFrameLengthRange() function. Thic CL implements the function for all audio encoders in WebRTC in preparation for making that function pure virtual in the interface. Bug: webrtc:11427 Change-Id: Ieab6b6c72c62af6ac9525a20fcb39bd477079551 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171503 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30890}
2020-03-24 16:00:51 +01:00
"../../api/units:time_delta",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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 //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
public_deps = [ ":g722_c" ] # no-presubmit-check TODO(webrtc:8603)
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
}
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",
]
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
deps = [ "../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",
Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. The WebRTC-SendSideBwe-WithOverhead field trial requires audio encoders to properly implement the AudioEncoder::GetFrameLengthRange() function. Thic CL implements the function for all audio encoders in WebRTC in preparation for making that function pure virtual in the interface. Bug: webrtc:11427 Change-Id: Ieab6b6c72c62af6ac9525a20fcb39bd477079551 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171503 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30890}
2020-03-24 16:00:51 +01:00
"../../api/units:time_delta",
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",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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 //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
public_deps = [ ":ilbc_c" ] # no-presubmit-check TODO(webrtc:8603)
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
}
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",
"../../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",
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
]
Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in audio_coding Bug: webrtc:12336 Change-Id: Icae229b957c2bfcc410788179a504c576cfde151 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201736 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32995}
2021-01-14 18:35:41 +01:00
absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
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("isac_vad") {
Reland "[ACM] iSAC audio codec removed" This is a reland of commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1 Original change's description: > [ACM] iSAC audio codec removed > > Note: this CL has to leave behind one part of iSAC, which is its VAD > currently used by AGC1 in APM. The target visibility has been > restricted and the VAD will be removed together with AGC1 when the > time comes. > > Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319 > > Bug: webrtc:14450 > Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421 > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38652} Bug: webrtc:14450 Change-Id: Ia22c4d7724b6022238235fede93e36e570a49376 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283843 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38665}
2022-11-11 16:52:46 +01:00
visibility += [ "../audio_processing/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
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
]
}
Reland "[ACM] iSAC audio codec removed" This is a reland of commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1 Original change's description: > [ACM] iSAC audio codec removed > > Note: this CL has to leave behind one part of iSAC, which is its VAD > currently used by AGC1 in APM. The target visibility has been > restricted and the VAD will be removed together with AGC1 when the > time comes. > > Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319 > > Bug: webrtc:14450 > Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421 > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38652} Bug: webrtc:14450 Change-Id: Ia22c4d7724b6022238235fede93e36e570a49376 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283843 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38665}
2022-11-11 16:52:46 +01:00
rtc_source_set("isac_bwinfo") {
sources = [ "codecs/isac/bandwidth_info.h" ]
deps = []
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",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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",
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
]
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
public_deps = [ ":pcm16b_c" ] # no-presubmit-check TODO(webrtc:8603)
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
}
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",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_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
"//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",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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 //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
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_numerics",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
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 //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../rtc_base:stringutils",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
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",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
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
]
iSAC unit test: test encode/decode via API wrapper Unit test to test the iSAC webrtc API wrapper, plus a minor change in the c iSAC wrapper. Bug: webrtc:10584 Change-Id: Iecbf6f3e7db5b3bdba41f8428254ae6a6a73e24a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168492 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30514}
2020-02-13 11:21:58 +01:00
public_deps = # no-presubmit-check TODO(webrtc:8603)
[ ":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
}
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",
Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. The WebRTC-SendSideBwe-WithOverhead field trial requires audio encoders to properly implement the AudioEncoder::GetFrameLengthRange() function. Thic CL implements the function for all audio encoders in WebRTC in preparation for making that function pure virtual in the interface. Bug: webrtc:11427 Change-Id: Ieab6b6c72c62af6ac9525a20fcb39bd477079551 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171503 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30890}
2020-03-24 16:00:51 +01:00
"../../api/units:time_delta",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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
"../../rtc_base:checks",
"../../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
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
"../../rtc_base:safe_minmax",
"../../rtc_base:stringutils",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_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
"//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",
]
iSAC unit test: test encode/decode via API wrapper Unit test to test the iSAC webrtc API wrapper, plus a minor change in the c iSAC wrapper. Bug: webrtc:10584 Change-Id: Iecbf6f3e7db5b3bdba41f8428254ae6a6a73e24a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168492 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30514}
2020-02-13 11:21:58 +01:00
public_deps = # no-presubmit-check TODO(webrtc:8603)
[ ":webrtc_opus_wrapper" ]
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
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
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 = [
Avoiding the noise pumping during DTX regions by just forwarding the refresh DTX packets that decrease the comfort noise level at the decoder. Bug: webrtc:12380 Change-Id: I60e4684150cb4880224f402a9bf42a72811863b3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202920 Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33174}
2021-02-05 09:05:46 +01:00
"../../api:array_view",
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",
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
]
Don't add libopus to public_deps, its headers are only used directly Bug: webrtc:8603 Change-Id: I2ce1f96a80dd23e420b3693b899d2b14382fd2d7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266765 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37363}
2022-06-28 09:21:44 +02:00
if (rtc_build_opus) {
deps += [ rtc_opus_dir ]
public_configs = [ "//third_party/opus:opus_config" ]
} 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
}
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
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
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 += [ "*" ]
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
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",
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//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",
Add a simple frame length controller. This will be used when adaptivePtime is enabled. Bug: chromium:1086942 Change-Id: I63c947c53a8c5b8e0825b78b847c3f7900197d6c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177421 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31544}
2020-06-18 15:48:39 +02:00
"audio_network_adaptor/frame_length_controller_v2.cc",
"audio_network_adaptor/frame_length_controller_v2.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
iSAC unit test: test encode/decode via API wrapper Unit test to test the iSAC webrtc API wrapper, plus a minor change in the c iSAC wrapper. Bug: webrtc:10584 Change-Id: Iecbf6f3e7db5b3bdba41f8428254ae6a6a73e24a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168492 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30514}
2020-02-13 11:21:58 +01:00
public_deps = # no-presubmit-check TODO(webrtc:8603)
[ ":audio_network_adaptor_config" ]
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
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 //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
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",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
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",
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
]
Add a simple frame length controller. This will be used when adaptivePtime is enabled. Bug: chromium:1086942 Change-Id: I63c947c53a8c5b8e0825b78b847c3f7900197d6c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177421 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31544}
2020-06-18 15:48:39 +02:00
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
Adopt absl::string_view in modules/audio_coding/ Bug: webrtc:13579 Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37573}
2022-07-20 12:53:07 +02:00
"//third_party/abseil-cpp/absl/strings",
Add a simple frame length controller. This will be used when adaptivePtime is enabled. Bug: chromium:1086942 Change-Id: I63c947c53a8c5b8e0825b78b847c3f7900197d6c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177421 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31544}
2020-06-18 15:48:39 +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/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",
Introduce a stable playout delay mode for NetEq. A packet arrival history is used to store the timing of incoming packets and tracks the earliest and latest packets by taking the difference between rtp timestamp and arrival time. The history is windowed to 2 seconds by default. The packet arrival history will replace the relative arrival delay tracker in a follow up cl. The playout delay is estimated by taking the difference between the current playout timestamp and the earliest packet arrival in the history. This method works better when DTX is used compared to the buffer level filter that it replaces. The threshold for acceleration is changed to be the maximum of the target delay and the maximum packet arrival delay in the history. This prevents any acceleration immediately after an underrun and gives some time to adapt the target delay to new network conditions. The logic when to decode the next packet after a packet loss is also changed to do concealment for the full loss duration unless the delay is too high. The new mode is default disabled and can be enabled using a field trial. Bug: webrtc:13322,webrtc:13966 Change-Id: Idfa0020584591261475b9ca350cc7c6531de9911 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259820 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36899}
2022-05-16 15:28:22 +02:00
"neteq/packet_arrival_history.cc",
"neteq/packet_arrival_history.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",
Improve adaptation to reordered packets in delay manager. This is done by adding a reorder optimizer that estimates the probability of receiving reordered packets. The optimal delay is decided by balancing the cost of increasing the delay against the probability of missing a reordered packet, resulting in a loss. This balance is decided using the `ms_per_loss_percent` parameter. The usage and parameters can be controlled via field trial. Bug: webrtc:10178 Change-Id: Ic484df0412af35610e74b3a6070f2bac7a926a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231541 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34954}
2021-09-08 16:35:50 +02:00
"neteq/reorder_optimizer.cc",
"neteq/reorder_optimizer.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",
Refactor delay manager. Split out `RelativeArrivalDelayTracker` and `DelayOptimizer` logic. This is in preparation for adding another `DelayOptimizer` specialized in handling reordered packets. Bug: webrtc:10178 Change-Id: Id3c1746d91980b171fa524f9b2b71cf11fc75f64 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231224 Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34938}
2021-09-07 14:24:56 +02:00
"neteq/underrun_optimizer.cc",
"neteq/underrun_optimizer.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",
[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",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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 event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../../rtc_base:event_tracer",
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",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
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",
Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I03b78bd2e411e9bcca199f85e4457511826cd17e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176745 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31649}
2020-07-07 15:53:34 +02:00
"../../rtc_base/synchronization:mutex",
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",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
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
Add a CreateNetEq method that takes an AudioDecoderFactory The NetEqFactory is currently expected to wrap the AudioDecoderFactory, but this turns out not to be a good idea. Instead, it makes more sense to pass the AudioDecoderFactory through the CreateNetEq method. Bug: webrtc:11005 Change-Id: I8027ff6593f40c92072e7e88157631dcf329a984 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160644 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29918}
2019-11-26 12:29:05 +01:00
rtc_source_set("default_neteq_factory") {
visibility += webrtc_default_visibility
sources = [
"neteq/default_neteq_factory.cc",
"neteq/default_neteq_factory.h",
]
deps = [
":neteq",
"../../api:scoped_refptr",
"../../api/audio_codecs:audio_codecs_api",
"../../api/neteq:default_neteq_controller_factory",
"../../api/neteq:neteq_api",
"../../system_wrappers:system_wrappers",
]
}
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 = [
Clean up the NetEqFactory API. This CL decouples NetEqFactory and AudioDecoderFactory. AudioDecoderFactory is used in more places than just inside of NetEq, so decoupling these makes sense. Bug: webrtc:11005 Change-Id: I78dd856e4248e398e69a65816b062ef30555b055 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161005 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29961}
2019-11-28 14:07:14 +01:00
":default_neteq_factory",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
":neteq",
Avoid using legacy rtp parser in neteq test::Packet Bug: None Change-Id: I9184954d9c99f0a34ae335d03843171864071e5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222648 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34316}
2021-06-16 14:23:22 +02:00
"../../api:array_view",
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",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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 copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../../rtc_base:copy_on_write_buffer",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../rtc_base:stringutils",
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",
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",
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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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 = [
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_tools",
":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
":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",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../rtc_base:buffer",
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",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../../rtc_base:ssl",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../rtc_base:stringutils",
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",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
Adopt absl::string_view in modules/audio_coding/ Bug: webrtc:13579 Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37573}
2022-07-20 12:53:07 +02:00
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//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
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 = [
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_input_audio_tools",
":neteq_tools_minimal",
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 //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../rtc_base:stringutils",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
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",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
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
]
}
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
]
Adopt absl::string_view in modules/audio_coding/ Bug: webrtc:13579 Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37573}
2022-07-20 12:53:07 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
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
}
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",
Stop using public_deps in logging/. Bug: webrtc:8603 Change-Id: Id0df997620a27e47067e4b21e4e8db16aec90640 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30940 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21228}
2017-12-12 12:46:13 +01:00
"../rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) Reason for revert: Fixing and re-landing. Original issue's description: > Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) > > Reason for revert: > ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here. > #include "webrtc/base/checks.h" > ^------------------- > The target: > //webrtc/test:fileutils_unittests > is including a file from the target: > //webrtc/base:rtc_base_approved > > It's usually best to depend directly on the destination target. > In some cases, the destination target is considered a subcomponent > of an intermediate target. In this case, the intermediate target > should depend publicly on the destination to forward the ability > to include headers. > > Dependency chain (there may also be others): > //webrtc/test:fileutils_unittests --> > //webrtc/test:fileutils --[private]--> > //webrtc/base:rtc_base_approved > > > Original issue's description: > > Enabling `gn check` on webrtc/test > > > > BUG=webrtc:6828 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2911203002 > > Cr-Commit-Position: refs/heads/master@{#18372} > > Committed: https://chromium.googlesource.com/external/webrtc/+/db5bb404b0f42a7c0a43f882b34ba1325d8cbae2 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6828 > > Review-Url: https://codereview.webrtc.org/2920763002 > Cr-Commit-Position: refs/heads/master@{#18375} > Committed: https://chromium.googlesource.com/external/webrtc/+/1a6f143d077aec85d919c516b4f74ada8b8c21a8 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2918793002 Cr-Commit-Position: refs/heads/master@{#18376}
2017-06-01 04:47:20 -07:00
]
Adopt absl::string_view in modules/audio_coding/ Bug: webrtc:13579 Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37573}
2022-07-20 12:53:07 +02:00
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
iSAC unit test: test encode/decode via API wrapper Unit test to test the iSAC webrtc API wrapper, plus a minor change in the c iSAC wrapper. Bug: webrtc:10584 Change-Id: Iecbf6f3e7db5b3bdba41f8428254ae6a6a73e24a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168492 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30514}
2020-02-13 11:21:58 +01:00
public_deps = # no-presubmit-check TODO(webrtc:8603)
[ "../../logging:rtc_event_log_proto" ]
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
}
Expose NetEqDecodingTest for re-use in chromium tests. This CL allows to trigger related tests when rolling opus (at chromium side). Namely: * TestOpusBitExactness * TestOpusDtxBitExactness This CL also prevents name clash for OpusTest: * modules/audio_coding/test/opus_test.h: Helper class. * modules/audio_coding/neteq/opus_unittest.cc: Local test fixture. Bug: chromium:1002973 Change-Id: If8470b5f64fbdb1f7a84b838bde62d8c90390f2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159033 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29759}
2019-11-11 18:05:42 +01:00
# Only used for test purpose. Since we want to use it from chromium
# (see audio_coding_modules_tests_shared below), we cannot guard it
# under rtc_include_tests.
proto_library("neteq_unittest_proto") {
testonly = true
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
sources = [ "neteq/neteq_unittest.proto" ]
Expose NetEqDecodingTest for re-use in chromium tests. This CL allows to trigger related tests when rolling opus (at chromium side). Namely: * TestOpusBitExactness * TestOpusDtxBitExactness This CL also prevents name clash for OpusTest: * modules/audio_coding/test/opus_test.h: Helper class. * modules/audio_coding/neteq/opus_unittest.cc: Local test fixture. Bug: chromium:1002973 Change-Id: If8470b5f64fbdb1f7a84b838bde62d8c90390f2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159033 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29759}
2019-11-11 18:05:42 +01:00
proto_out_dir = "modules/audio_coding/neteq"
}
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
}
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 = [
Expose NetEqDecodingTest for re-use in chromium tests. This CL allows to trigger related tests when rolling opus (at chromium side). Namely: * TestOpusBitExactness * TestOpusDtxBitExactness This CL also prevents name clash for OpusTest: * modules/audio_coding/test/opus_test.h: Helper class. * modules/audio_coding/neteq/opus_unittest.cc: Local test fixture. Bug: chromium:1002973 Change-Id: If8470b5f64fbdb1f7a84b838bde62d8c90390f2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159033 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29759}
2019-11-11 18:05:42 +01:00
"neteq/test/neteq_decoding_test.cc",
"neteq/test/neteq_decoding_test.h",
"neteq/test/result_sink.cc",
"neteq/test/result_sink.h",
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
"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",
Clean up the NetEqFactory API. This CL decouples NetEqFactory and AudioDecoderFactory. AudioDecoderFactory is used in more places than just inside of NetEq, so decoupling these makes sense. Bug: webrtc:11005 Change-Id: I78dd856e4248e398e69a65816b062ef30555b055 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161005 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29961}
2019-11-28 14:07:14 +01:00
":default_neteq_factory",
Expose NetEqDecodingTest for re-use in chromium tests. This CL allows to trigger related tests when rolling opus (at chromium side). Namely: * TestOpusBitExactness * TestOpusDtxBitExactness This CL also prevents name clash for OpusTest: * modules/audio_coding/test/opus_test.h: Helper class. * modules/audio_coding/neteq/opus_unittest.cc: Local test fixture. Bug: chromium:1002973 Change-Id: If8470b5f64fbdb1f7a84b838bde62d8c90390f2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159033 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29759}
2019-11-11 18:05:42 +01:00
":neteq_test_tools",
":neteq_tools_minimal",
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
":webrtc_opus_wrapper",
"..:module_api",
Expose NetEqDecodingTest for re-use in chromium tests. This CL allows to trigger related tests when rolling opus (at chromium side). Namely: * TestOpusBitExactness * TestOpusDtxBitExactness This CL also prevents name clash for OpusTest: * modules/audio_coding/test/opus_test.h: Helper class. * modules/audio_coding/neteq/opus_unittest.cc: Local test fixture. Bug: chromium:1002973 Change-Id: If8470b5f64fbdb1f7a84b838bde62d8c90390f2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159033 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29759}
2019-11-11 18:05:42 +01:00
"../../api:rtp_headers",
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
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/audio_codecs:builtin_audio_encoder_factory",
Expose NetEqDecodingTest for re-use in chromium tests. This CL allows to trigger related tests when rolling opus (at chromium side). Namely: * TestOpusBitExactness * TestOpusDtxBitExactness This CL also prevents name clash for OpusTest: * modules/audio_coding/test/opus_test.h: Helper class. * modules/audio_coding/neteq/opus_unittest.cc: Local test fixture. Bug: chromium:1002973 Change-Id: If8470b5f64fbdb1f7a84b838bde62d8c90390f2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159033 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29759}
2019-11-11 18:05:42 +01:00
"../../api/neteq:neteq_api",
"../../rtc_base",
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
"../../rtc_base:checks",
Expose NetEqDecodingTest for re-use in chromium tests. This CL allows to trigger related tests when rolling opus (at chromium side). Namely: * TestOpusBitExactness * TestOpusDtxBitExactness This CL also prevents name clash for OpusTest: * modules/audio_coding/test/opus_test.h: Helper class. * modules/audio_coding/neteq/opus_unittest.cc: Local test fixture. Bug: chromium:1002973 Change-Id: If8470b5f64fbdb1f7a84b838bde62d8c90390f2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159033 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29759}
2019-11-11 18:05:42 +01:00
"../../rtc_base:ignore_wundef",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../../rtc_base:ssl",
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
"../../rtc_base:stringutils",
Expose NetEqDecodingTest for re-use in chromium tests. This CL allows to trigger related tests when rolling opus (at chromium side). Namely: * TestOpusBitExactness * TestOpusDtxBitExactness This CL also prevents name clash for OpusTest: * modules/audio_coding/test/opus_test.h: Helper class. * modules/audio_coding/neteq/opus_unittest.cc: Local test fixture. Bug: chromium:1002973 Change-Id: If8470b5f64fbdb1f7a84b838bde62d8c90390f2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159033 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29759}
2019-11-11 18:05:42 +01:00
"../../system_wrappers",
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
"../../test:fileutils",
"../../test:test_support",
Expose NetEqDecodingTest for re-use in chromium tests. This CL allows to trigger related tests when rolling opus (at chromium side). Namely: * TestOpusBitExactness * TestOpusDtxBitExactness This CL also prevents name clash for OpusTest: * modules/audio_coding/test/opus_test.h: Helper class. * modules/audio_coding/neteq/opus_unittest.cc: Local test fixture. Bug: chromium:1002973 Change-Id: If8470b5f64fbdb1f7a84b838bde62d8c90390f2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159033 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29759}
2019-11-11 18:05:42 +01:00
"//testing/gtest",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
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
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
defines = audio_coding_defines
Expose NetEqDecodingTest for re-use in chromium tests. This CL allows to trigger related tests when rolling opus (at chromium side). Namely: * TestOpusBitExactness * TestOpusDtxBitExactness This CL also prevents name clash for OpusTest: * modules/audio_coding/test/opus_test.h: Helper class. * modules/audio_coding/neteq/opus_unittest.cc: Local test fixture. Bug: chromium:1002973 Change-Id: If8470b5f64fbdb1f7a84b838bde62d8c90390f2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159033 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29759}
2019-11-11 18:05:42 +01:00
if (rtc_enable_protobuf) {
defines += [ "WEBRTC_NETEQ_UNITTEST_BITEXACT" ]
deps += [ ":neteq_unittest_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
}
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 = [
":audio_encoder_cng",
":g711",
":g722",
":pcm16b",
Suppress C5041 constexpr warning for MSVC 2019 Disable the C5041 warning which makes the build fail. This is a C++17-only change and WebRTC doesn't support C++17 yet, so the code is technically correct, but fails to build on MSVC 2019 and warning-as-error active. Also fix another warning-as-error build error with MSVC 2019 due to ignoring the result of a [[nodiscard]] function. No-Presubmit: True Bug: webrtc:11275,webrtc:11276 Change-Id: I891a894ee87252f96e84fd8d282576f46907256f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165781 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30244}
2020-01-13 11:58:26 +00:00
"../../common_audio",
"../../system_wrappers",
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
]
if (rtc_include_ilbc) {
audio_coding_deps += [ ":ilbc" ]
}
if (rtc_include_opus) {
audio_coding_deps += [ ":webrtc_opus" ]
}
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",
]
}
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
if (!build_with_chromium) {
group("audio_coding_tests") {
visibility += webrtc_default_visibility
testonly = true
public_deps = [ # no-presubmit-check TODO(webrtc:8603)
":acm_receive_test",
":acm_send_test",
":audio_codec_speed_tests",
":audio_decoder_unittests",
":audio_decoder_unittests",
":g711_test",
":g722_test",
":ilbc_test",
":neteq_ilbc_quality_test",
":neteq_opus_quality_test",
":neteq_pcm16b_quality_test",
":neteq_pcmu_quality_test",
":neteq_speed_test",
":rtp_analyze",
":rtp_encode",
":rtp_jitter",
":rtpcat",
":webrtc_opus_fec_test",
]
if (rtc_enable_protobuf) {
public_deps += # no-presubmit-check TODO(webrtc:8603)
[ ":neteq_rtpplay" ]
}
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
}
}
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/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/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 //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../rtc_base:stringutils",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I03b78bd2e411e9bcca199f85e4457511826cd17e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176745 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31649}
2020-07-07 15:53:34 +02:00
"../../rtc_base/synchronization:mutex",
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",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 5/inf Add field trials to audio api. It is added as a pointer with nullptr as default. It is not (yet) used anywhere. Usage of field trials comes in subsequent patches. Bug: webrtc:10335 Change-Id: Icbe22d95c356a6fefde34590f11ea63f005ab09e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255521 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36213}
2022-03-15 14:29:00 +01:00
"../../test:scoped_key_value_config",
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",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
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",
Migrate audio perf tests on new perf metrics export API Bug: b/246095034 Change-Id: Id659e43c116428cab47d334c93a6036f74dbb8e1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276626 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38192}
2022-09-25 00:29:43 +02:00
"../../api/test/metrics:global_metrics_logger_and_exporter",
"../../api/test/metrics:metric",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
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 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
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
deps = [
":audio_coding",
":neteq_tools",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_tools_minimal",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../../api:scoped_refptr",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../test:test_support",
"//testing/gtest",
]
deps += audio_coding_deps
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//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
}
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
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
deps = [
":audio_coding",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_input_audio_tools",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
":neteq_tools",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_tools_minimal",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/audio_codecs:builtin_audio_encoder_factory",
"../../rtc_base:checks",
"../../rtc_base:stringutils",
"../../test:test_support",
"//testing/gtest",
]
deps += audio_coding_deps
Add absl_deps field for rtc_test and rtc_executable To be able to build these targets in chromium we need to replace all abseil dependencies with "//third_party/abseil-cpp:absl". Bug: webrtc:12404 Change-Id: Ie0f6af73f2abc73e5744520cfd9a6414e2f948e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202762 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33108}
2021-01-29 10:50:19 +00:00
absl_deps = [ "//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
}
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
if (!build_with_chromium) {
audio_decoder_unittests_resources =
[ "../../resources/audio_coding/testfile32kHz.pcm" ]
GN: Fix resource files for iOS test target In order to get resource files to be properly packaged into the .app for a unit test on iOS, the resource files needs to be listed as sources in a bundle_data target. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2292853002 Cr-Commit-Position: refs/heads/master@{#13968}
2016-08-30 02:53:49 -07:00
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
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: 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
}
GN conversion of audio_decoder_unittests BUG=webrtc 6036 Review-Url: https://codereview.webrtc.org/2180853006 Cr-Commit-Position: refs/heads/master@{#13588}
2016-08-01 07:49:42 -07:00
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_test("audio_decoder_unittests") {
testonly = true
sources = [ "neteq/audio_decoder_unittest.cc" ]
GN conversion of audio_decoder_unittests BUG=webrtc 6036 Review-Url: https://codereview.webrtc.org/2180853006 Cr-Commit-Position: refs/heads/master@{#13588}
2016-08-01 07:49:42 -07:00
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
defines = neteq_defines
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [
":ilbc",
":neteq",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_input_audio_tools",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
":neteq_tools",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../common_audio",
"../../rtc_base/system:arch",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
"../../test:fileutils",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../test:test_main",
"../../test:test_support",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
"//testing/gtest",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
] + audio_coding_deps
data = audio_decoder_unittests_resources
if (is_android) {
Use WebRTC's Java VM initialization in tests. WebRTC should not depend on chromium's //base. Bug: webrtc:13662 Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36581}
2022-04-19 15:32:37 +02:00
use_default_launcher = false
deps += [
"//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
"//testing/android/native_test:native_test_java",
"//testing/android/native_test:native_test_support",
]
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
shard_timeout = 900
}
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) {
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 = [
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_input_audio_tools",
":neteq_tools",
":neteq_tools_minimal",
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 //rtc_base:refcount from public deps Bug: webrtc:8603 Change-Id: Ib27a107ae809df739492846175f0e9c4af40d21a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257910 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36447}
2022-04-05 02:54:12 +02:00
"../../rtc_base:refcount",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
]
Adopt absl::string_view in modules/audio_coding/ Bug: webrtc:13579 Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37573}
2022-07-20 12:53:07 +02:00
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
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",
Move test utilities into more specific build targets Move audio- and video-specific utilities to audio_test_common (newly added target) and video_test_common. Bug: webrtc:10198 Change-Id: Ia10fa5c0a51d9b1f37db4964984d22fc5b269bb1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268980 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37570}
2022-07-20 10:20:33 +02:00
"../../test:audio_test_common",
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
]
}
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
}
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
if (rtc_enable_protobuf && !build_with_chromium) {
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",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_tools_minimal",
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
]
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
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
}
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
if (!build_with_chromium) {
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",
]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
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}}" ]
}
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
}
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_test("audio_codec_speed_tests") {
testonly = true
defines = []
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
deps = [
"../../rtc_base:macromagic",
"../../test:fileutils",
]
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [
"codecs/opus/opus_speed_test.cc",
"codecs/tools/audio_codec_speed_test.cc",
"codecs/tools/audio_codec_speed_test.h",
]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
data = audio_codec_speed_tests_resources
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
if (is_android) {
Use WebRTC's Java VM initialization in tests. WebRTC should not depend on chromium's //base. Bug: webrtc:13662 Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36581}
2022-04-19 15:32:37 +02:00
use_default_launcher = false
deps += [
"//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
"//testing/android/native_test:native_test_java",
"//testing/android/native_test:native_test_support",
]
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
if (is_ios) {
deps += [ ":audio_codec_speed_tests_data" ]
}
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps += [
":webrtc_opus",
Replace assert() with RTC_DCHECK(). CL partially auto-generated with: git grep -l "\bassert(" | grep "\.[c|h]" | \ xargs sed -i 's/\bassert(/RTC_DCHECK(/g' And with: git grep -l "RTC_DCHECK(false)" | \ xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g' With some manual changes to include "rtc_base/checks.h" where needed. A follow-up CL will remove assert() from Obj-C code as well and remove the #include of <assert.h>. The choice to replace with RTC_DCHECK is because assert() is because RTC_DCHECK has similar behavior as assert() based on NDEBUG. This CL also contains manual changes to switch from basic RTC_DCHECK to other (preferred) versions like RTC_DCHECK_GT (and similar). Bug: webrtc:6779 Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34442}
2021-07-08 20:08:20 +02:00
"../../rtc_base:checks",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../test:test_main",
"../../test:test_support",
"../audio_processing",
"//testing/gtest",
]
}
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
}
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 = [
Clean up the NetEqFactory API. This CL decouples NetEqFactory and AudioDecoderFactory. AudioDecoderFactory is used in more places than just inside of NetEq, so decoupling these makes sense. Bug: webrtc:11005 Change-Id: I78dd856e4248e398e69a65816b062ef30555b055 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161005 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29961}
2019-11-28 14:07:14 +01:00
":default_neteq_factory",
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
":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",
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",
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",
]
}
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
if (!build_with_chromium) {
rtc_library("neteq_quality_test_support") {
testonly = true
sources = [
"neteq/tools/neteq_quality_test.cc",
"neteq/tools/neteq_quality_test.h",
]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [
":default_neteq_factory",
":neteq",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_input_audio_tools",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
":neteq_test_tools",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_tools_minimal",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/neteq:neteq_api",
"../../rtc_base:checks",
Make it easier to specify in/out files for neteq_quality_test. Bug: b/251155608 Change-Id: I174351d76a83de651f5ef025606712333a83cf52 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278786 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38358}
2022-10-11 19:22:00 +00:00
"../../rtc_base:stringutils",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../system_wrappers",
"../../test:fileutils",
"../../test:test_support",
"//testing/gtest",
]
Adopt absl::string_view in modules/audio_coding/ Bug: webrtc:13579 Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37573}
2022-07-20 12:53:07 +02:00
absl_deps = [
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/strings",
]
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
}
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_executable("rtp_encode") {
testonly = true
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
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
deps = [
":audio_coding",
":audio_encoder_cng",
":neteq_input_audio_tools",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs/L16:audio_encoder_L16",
"../../api/audio_codecs/g711:audio_encoder_g711",
"../../api/audio_codecs/g722:audio_encoder_g722",
"../../api/audio_codecs/ilbc:audio_encoder_ilbc",
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../rtc_base:safe_conversions",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
"//third_party/abseil-cpp/absl/memory",
]
deps += audio_coding_deps
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [ "neteq/tools/rtp_encode.cc" ]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_executable("rtp_jitter") {
testonly = true
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
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
deps = [
"../../api:array_view",
"../../rtc_base:buffer",
"../rtp_rtcp:rtp_rtcp_format",
]
deps += audio_coding_deps
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [ "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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_executable("rtpcat") {
testonly = true
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [ "neteq/tools/rtpcat.cc" ]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [
"../../rtc_base:checks",
"../../test:rtp_test_utils",
"//testing/gtest",
]
}
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_executable("rtp_analyze") {
testonly = true
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [ "neteq/tools/rtp_analyze.cc" ]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [
":neteq",
":neteq_test_tools",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_tools_minimal",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
":pcm16b",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
]
}
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_executable("neteq_opus_quality_test") {
testonly = true
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [ "neteq/test/neteq_opus_quality_test.cc" ]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [
":neteq",
":neteq_quality_test_support",
":neteq_tools",
":webrtc_opus",
"../../test:test_main",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
]
}
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_executable("neteq_speed_test") {
testonly = true
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [ "neteq/test/neteq_speed_test.cc" ]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [
":neteq",
":neteq_test_support",
"../../rtc_base:checks",
"../../test:test_support",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
]
}
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_executable("neteq_ilbc_quality_test") {
testonly = true
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [ "neteq/test/neteq_ilbc_quality_test.cc" ]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [
":ilbc",
":neteq",
":neteq_quality_test_support",
":neteq_tools",
"../../rtc_base:checks",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../test:fileutils",
"../../test:test_main",
"//testing/gtest",
"//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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_executable("neteq_pcmu_quality_test") {
testonly = true
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [ "neteq/test/neteq_pcmu_quality_test.cc" ]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [
":g711",
":neteq",
":neteq_quality_test_support",
"../../rtc_base:checks",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../test:fileutils",
"../../test:test_main",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
]
}
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_executable("neteq_pcm16b_quality_test") {
testonly = true
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [ "neteq/test/neteq_pcm16b_quality_test.cc" ]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [
":neteq",
":neteq_quality_test_support",
":pcm16b",
"../../rtc_base:checks",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../test:fileutils",
"../../test:test_main",
"//testing/gtest",
"//third_party/abseil-cpp/absl/flags:flag",
]
}
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
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
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
sources = [ "codecs/g711/test/testG711.cc" ]
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
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
deps = [ ":g711" ]
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
}
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
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
sources = [ "codecs/g722/test/testG722.cc" ]
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
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
deps = [ ":g722" ]
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
}
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
if (!build_with_chromium) {
rtc_executable("ilbc_test") {
testonly = true
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [ "codecs/ilbc/test/iLBC_test.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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [ ":ilbc" ]
}
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_executable("webrtc_opus_fec_test") {
testonly = true
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [ "codecs/opus/opus_fec_test.cc" ]
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [
":webrtc_opus",
"../../common_audio",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../test:fileutils",
"../../test:test_main",
"../../test:test_support",
"//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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
rtc_library("audio_coding_unittests") {
testonly = true
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
sources = [
"acm2/acm_receiver_unittest.cc",
"acm2/acm_remixing_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",
"audio_network_adaptor/event_log_writer_unittest.cc",
"audio_network_adaptor/fec_controller_plr_based_unittest.cc",
"audio_network_adaptor/frame_length_controller_unittest.cc",
"audio_network_adaptor/frame_length_controller_v2_unittest.cc",
"audio_network_adaptor/util/threshold_curve_unittest.cc",
"codecs/builtin_audio_decoder_factory_unittest.cc",
"codecs/builtin_audio_encoder_factory_unittest.cc",
"codecs/cng/audio_encoder_cng_unittest.cc",
"codecs/cng/cng_unittest.cc",
"codecs/ilbc/ilbc_unittest.cc",
"codecs/legacy_encoded_audio_frame_unittest.cc",
"codecs/opus/audio_decoder_multi_channel_opus_unittest.cc",
"codecs/opus/audio_encoder_multi_channel_opus_unittest.cc",
"codecs/opus/audio_encoder_opus_unittest.cc",
"codecs/opus/opus_bandwidth_unittest.cc",
"codecs/opus/opus_unittest.cc",
"codecs/red/audio_encoder_copy_red_unittest.cc",
"neteq/audio_multi_vector_unittest.cc",
"neteq/audio_vector_unittest.cc",
"neteq/background_noise_unittest.cc",
"neteq/buffer_level_filter_unittest.cc",
"neteq/comfort_noise_unittest.cc",
"neteq/decision_logic_unittest.cc",
"neteq/decoder_database_unittest.cc",
"neteq/delay_manager_unittest.cc",
"neteq/dsp_helper_unittest.cc",
"neteq/dtmf_buffer_unittest.cc",
"neteq/dtmf_tone_generator_unittest.cc",
"neteq/expand_unittest.cc",
"neteq/histogram_unittest.cc",
"neteq/merge_unittest.cc",
"neteq/mock/mock_buffer_level_filter.h",
"neteq/mock/mock_decoder_database.h",
"neteq/mock/mock_delay_manager.h",
"neteq/mock/mock_dtmf_buffer.h",
"neteq/mock/mock_dtmf_tone_generator.h",
"neteq/mock/mock_expand.h",
"neteq/mock/mock_histogram.h",
"neteq/mock/mock_neteq_controller.h",
"neteq/mock/mock_packet_buffer.h",
"neteq/mock/mock_red_payload_splitter.h",
"neteq/mock/mock_statistics_calculator.h",
"neteq/nack_tracker_unittest.cc",
"neteq/neteq_decoder_plc_unittest.cc",
"neteq/neteq_impl_unittest.cc",
"neteq/neteq_network_stats_unittest.cc",
"neteq/neteq_stereo_unittest.cc",
"neteq/neteq_unittest.cc",
"neteq/normal_unittest.cc",
Introduce a stable playout delay mode for NetEq. A packet arrival history is used to store the timing of incoming packets and tracks the earliest and latest packets by taking the difference between rtp timestamp and arrival time. The history is windowed to 2 seconds by default. The packet arrival history will replace the relative arrival delay tracker in a follow up cl. The playout delay is estimated by taking the difference between the current playout timestamp and the earliest packet arrival in the history. This method works better when DTX is used compared to the buffer level filter that it replaces. The threshold for acceleration is changed to be the maximum of the target delay and the maximum packet arrival delay in the history. This prevents any acceleration immediately after an underrun and gives some time to adapt the target delay to new network conditions. The logic when to decode the next packet after a packet loss is also changed to do concealment for the full loss duration unless the delay is too high. The new mode is default disabled and can be enabled using a field trial. Bug: webrtc:13322,webrtc:13966 Change-Id: Idfa0020584591261475b9ca350cc7c6531de9911 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259820 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36899}
2022-05-16 15:28:22 +02:00
"neteq/packet_arrival_history_unittest.cc",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"neteq/packet_buffer_unittest.cc",
"neteq/post_decode_vad_unittest.cc",
"neteq/random_vector_unittest.cc",
"neteq/red_payload_splitter_unittest.cc",
Improve adaptation to reordered packets in delay manager. This is done by adding a reorder optimizer that estimates the probability of receiving reordered packets. The optimal delay is decided by balancing the cost of increasing the delay against the probability of missing a reordered packet, resulting in a loss. This balance is decided using the `ms_per_loss_percent` parameter. The usage and parameters can be controlled via field trial. Bug: webrtc:10178 Change-Id: Ic484df0412af35610e74b3a6070f2bac7a926a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231541 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34954}
2021-09-08 16:35:50 +02:00
"neteq/reorder_optimizer_unittest.cc",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"neteq/statistics_calculator_unittest.cc",
"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",
Refactor delay manager. Split out `RelativeArrivalDelayTracker` and `DelayOptimizer` logic. This is in preparation for adding another `DelayOptimizer` specialized in handling reordered packets. Bug: webrtc:10178 Change-Id: Id3c1746d91980b171fa524f9b2b71cf11fc75f64 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231224 Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34938}
2021-09-07 14:24:56 +02:00
"neteq/underrun_optimizer_unittest.cc",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
]
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
deps = [
":acm_receive_test",
":acm_send_test",
":audio_coding",
":audio_coding_module_typedefs",
":audio_coding_modules_tests_shared",
":audio_coding_opus_common",
":audio_encoder_cng",
":audio_network_adaptor",
":default_neteq_factory",
":g711",
":ilbc",
":legacy_encoded_audio_frame",
":mocks",
":neteq",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_input_audio_tools",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
":neteq_test_support",
":neteq_test_tools",
Remove public_deps usage in neteq build targets. Bug: b/36882554 Change-Id: I9a020e534a9f2c93de09684865a5bdddc60bd55d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266762 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@google.com> Cr-Commit-Position: refs/heads/main@{#37323}
2022-06-24 14:23:17 +02:00
":neteq_tools",
":neteq_tools_minimal",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
":pcm16b",
":red",
":webrtc_cng",
":webrtc_opus",
"..:module_api",
"..:module_api_public",
"../../api:array_view",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/audio_codecs:builtin_audio_encoder_factory",
"../../api/audio_codecs/opus:audio_decoder_multiopus",
"../../api/audio_codecs/opus:audio_decoder_opus",
"../../api/audio_codecs/opus:audio_encoder_multiopus",
"../../api/audio_codecs/opus:audio_encoder_opus",
"../../api/neteq:default_neteq_controller_factory",
"../../api/neteq:neteq_api",
"../../api/neteq:neteq_controller_api",
"../../api/neteq:tick_timer",
"../../api/neteq:tick_timer_unittest",
"../../api/rtc_event_log",
"../../common_audio",
"../../common_audio:common_audio_c",
"../../common_audio:mock_common_audio",
"../../logging:mocks",
"../../logging:rtc_event_audio",
"../../modules/rtp_rtcp:rtp_rtcp_format",
"../../rtc_base",
"../../rtc_base:checks",
"../../rtc_base:ignore_wundef",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
Remove platform_thread from //rtc_base:rtc_base_approved public_deps While the target has a restricted visibility, since it was in rtc_base_approved public deps, a lot of targets were able to bypass the visibility check. So we remove the visibility restrictions and use the dependency explicitely everywhere instead. Bug: webrtc:8603 Change-Id: I94a03fdf7f94c54ab72081a58dd648e2cca73d17 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258944 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36566}
2022-04-14 12:18:28 +02:00
"../../rtc_base:platform_thread",
Remove //rtc_base:refcount from public deps Bug: webrtc:8603 Change-Id: Ib27a107ae809df739492846175f0e9c4af40d21a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257910 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36447}
2022-04-05 02:54:12 +02:00
"../../rtc_base:refcount",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../rtc_base:rtc_base_tests_utils",
Remove //rtc_base:rtc_event from public deps Bug: webrtc:8603 Change-Id: Ib99f43043da17723c939b0fe2aa9f3e515462c93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257911 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36450}
2022-04-04 16:57:52 +02:00
"../../rtc_base:rtc_event",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../rtc_base:sanitizer",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../../rtc_base:ssl",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../rtc_base:stringutils",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../rtc_base:timeutils",
"../../rtc_base/synchronization:mutex",
"../../rtc_base/system:arch",
"../../system_wrappers",
"../../test:audio_codec_mocks",
Move test utilities into more specific build targets Move audio- and video-specific utilities to audio_test_common (newly added target) and video_test_common. Bug: webrtc:10198 Change-Id: Ia10fa5c0a51d9b1f37db4964984d22fc5b269bb1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268980 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37570}
2022-07-20 10:20:33 +02:00
"../../test:audio_test_common",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../test:field_trial",
"../../test:fileutils",
"../../test:rtc_expect_death",
"../../test:rtp_test_utils",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 7/inf Convert audio/ and collateral (audio encoder copy red). Bug: webrtc:10335 Change-Id: Iac54c0cfd2f62f4402f3deec35ae2725ec35b81a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255820 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36229}
2022-03-16 13:50:58 +01:00
"../../test:scoped_key_value_config",
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
"../../test:test_common",
"../../test:test_support",
"codecs/opus/test",
"codecs/opus/test:test_unittest",
"//testing/gtest",
]
absl_deps = [
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
defines = audio_coding_defines
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
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
if (rtc_enable_protobuf) {
defines += [ "WEBRTC_NETEQ_UNITTEST_BITEXACT" ]
deps += [
":ana_config_proto",
":neteq_unittest_proto",
]
}
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
}
}
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 += [ "*" ]
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
sources = [ "codecs/audio_decoder.h" ]
deps = [ "../../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
}
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 += [ "*" ]
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
sources = [ "codecs/audio_encoder.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
}
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