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webrtc_m130/webrtc/voice_engine/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 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
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
rtc_static_library("audio_coder") {
sources = [
"coder.cc",
"coder.h",
]
deps = [
"..:webrtc_common",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../api/audio_codecs:builtin_audio_decoder_factory",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../api/audio_codecs:builtin_audio_encoder_factory",
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
"../modules:module_api",
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
"../modules/audio_coding",
"../modules/audio_coding:audio_format_conversion",
"../modules/audio_coding:rent_a_codec",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_static_library("file_player") {
sources = [
"file_player.cc",
"file_player.h",
]
deps = [
":audio_coder",
"..:webrtc_common",
"../common_audio",
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
"../modules:module_api",
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
"../modules/media_file",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base_approved",
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_static_library("file_recorder") {
sources = [
"file_recorder.cc",
"file_recorder.h",
]
deps = [
":audio_coder",
"..:webrtc_common",
Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide: const int16_t* data() const; int16_t* mutable_data(); - data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames. - mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_. These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation. This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later. BUG=webrtc:7343 TBR=henrika Review-Url: https://codereview.webrtc.org/2750783004 Cr-Commit-Position: refs/heads/master@{#18543}
2017-06-12 12:45:32 -07:00
"../audio/utility:audio_frame_operations",
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
"../common_audio",
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
"../modules:module_api",
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
"../modules/media_file:media_file",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base_approved",
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
"../system_wrappers",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("voice_engine") {
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 = [
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
"channel.cc",
"channel.h",
"channel_manager.cc",
"channel_manager.h",
Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1459083007 Cr-Commit-Position: refs/heads/master@{#10788}
2015-11-25 08:16:52 -08:00
"channel_proxy.cc",
"channel_proxy.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
"include/voe_base.h",
"include/voe_codec.h",
"include/voe_errors.h",
"include/voe_file.h",
"include/voe_network.h",
"include/voe_rtp_rtcp.h",
"monitor_module.h",
"output_mixer.cc",
"output_mixer.h",
"shared_data.cc",
"shared_data.h",
"statistics.cc",
"statistics.h",
"transmit_mixer.cc",
"transmit_mixer.h",
Add TransportFeedbackPacketLossTracker. This CL is to calculate packet loss metrics from TransportFeedback. The outcome of this will be passed down to audio encoder. BUG=webrtc:6904 Review-Url: https://codereview.webrtc.org/2579613003 Cr-Commit-Position: refs/heads/master@{#16217}
2017-01-23 08:07:05 -08:00
"transport_feedback_packet_loss_tracker.cc",
"transport_feedback_packet_loss_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
"utility.cc",
"utility.h",
"voe_base_impl.cc",
"voe_base_impl.h",
"voe_codec_impl.cc",
"voe_codec_impl.h",
"voe_file_impl.cc",
"voe_file_impl.h",
"voe_network_impl.cc",
"voe_network_impl.h",
"voe_rtp_rtcp_impl.cc",
"voe_rtp_rtcp_impl.h",
"voice_engine_defines.h",
"voice_engine_impl.cc",
"voice_engine_impl.h",
]
if (is_win) {
defines = [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]
GN: Fix Windows build. This required a tiny include fix in src/third_party/winsdk_samples/src which was committed in https://code.google.com/p/webrtc/source/detail?r=7951 This incorporates contribution from vchigrin@yandex-team.ru in https://webrtc-codereview.appspot.com/29299004/ BUG=261,1348,4105 R=pbos@webrtc.org TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8027 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 21:22:01 +00:00
cflags = [
# TODO(kjellander): Bug 261: fix this warning.
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
"/wd4373", # Virtual function override.
GN: Fix Windows build. This required a tiny include fix in src/third_party/winsdk_samples/src which was committed in https://code.google.com/p/webrtc/source/detail?r=7951 This incorporates contribution from vchigrin@yandex-team.ru in https://webrtc-codereview.appspot.com/29299004/ BUG=261,1348,4105 R=pbos@webrtc.org TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8027 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 21:22:01 +00:00
]
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: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: 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: Move audio_coding to public_deps in voice engine audio_coding should be in public_deps, in order to a define used by voe_output_test.ccto be exported, as is done in GYP NOTRY=True BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2321783003 Cr-Commit-Position: refs/heads/master@{#14161}
2016-09-09 05:32:14 -07:00
public_deps = [
"../modules/audio_coding",
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
deps = [
Update formatting of AudioLevel class These changes are all no-op, only affecting the appearance of the code. The file names are changed to match the class name. BUG=none Review-Url: https://codereview.webrtc.org/2731993002 Cr-Commit-Position: refs/heads/master@{#17087}
2017-03-07 01:58:55 -08:00
":audio_level",
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
":file_player",
":file_recorder",
Fix style violations in common_types.h and config.h Mostly, it's about moving constructors and descructors to the .cc files, so that they won't be inlined everywhere. The reason this CL is so big is that a lot of code was using common_types.h without declaring a dependency on webrtc_common, which broke the build once common_types.h started to depend on common_types.cc. BUG=163 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26089004 Cr-Commit-Position: refs/heads/master@{#8516} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 13:59:22 +00:00
"..:webrtc_common",
Made AudioReceiveStream a mixer participant. Methods to facilitate this are added to ChannelProxy and voe::Channel. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2378143004 Cr-Commit-Position: refs/heads/master@{#14707}
2016-10-20 06:32:39 -07:00
"../api:audio_mixer_api",
Move webrtc/audio_*.h to webrtc/api/call BUG=webrtc:5878 NOTRY=True Review-Url: https://codereview.webrtc.org/2059703002 Cr-Commit-Position: refs/heads/master@{#13996}
2016-08-31 07:33:05 -07:00
"../api:call_api",
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) Reason for revert: Re-land, reverting did not fix bug. https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 Original issue's description: > Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) > > Reason for revert: > Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see > > https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 > > Original issue's description: > > Added the GetSources() to the RtpReceiverInterface and implemented > > it for the AudioRtpReceiver. > > > > This method returns a vector of RtpSource(both CSRC source and SSRC > > source) which contains the ID of a source, the timestamp, the source > > type (SSRC or CSRC) and the audio level. > > > > The RtpSource objects are buffered and maintained by the > > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called, > > the info of the contributing source will be pulled along the object > > chain: > > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel -> > > AudioReceiveStream -> voe::Channel -> RtpRtcp module > > > > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource > > > > BUG=chromium:703122 > > TBR=stefan@webrtc.org, danilchap@webrtc.org > > > > Review-Url: https://codereview.webrtc.org/2770233003 > > Cr-Commit-Position: refs/heads/master@{#17591} > > Committed: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053 > > TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=chromium:703122 > > Review-Url: https://codereview.webrtc.org/2809613002 > Cr-Commit-Position: refs/heads/master@{#17616} > Committed: https://chromium.googlesource.com/external/webrtc/+/fbcc5cb3869d1370008e40f24fc03ac8fb69c675 TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:703122 Review-Url: https://codereview.webrtc.org/2810623003 Cr-Commit-Position: refs/heads/master@{#17621}
2017-04-10 07:39:05 -07:00
"../api:libjingle_peerconnection_api",
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. transport.h defines an interface for sending rtp and rtcp packets, which is used by MediaChannel in webrtc/media/engine, {Audio|Video}{Send|Receive}Stream and in a few other places. It was part of the build target //webrtc:webrtc, which is a monolithic target with all webrtc production code. This CL moves the header to its own target in webrtc/api and deprecates the old location. Targets in webrtc/api should in general only depend on other targets in webrtc/api. The target webrtc/api:call_api depends on transport.h. This change also makes webrtc/voice_engine pass GN's header include checker and is needed in order for webrtc/api:call_api to pass it. transport.h will be completely removed in a follow-up CL in a few weeks after clients have updated their includes. NOTRY=True BUG=webrtc:5589, webrtc:5878, webrtc:6785 Review-Url: https://codereview.webrtc.org/2426563003 Cr-Commit-Position: refs/heads/master@{#15267}
2016-11-28 07:02:13 -08:00
"../api:transport_api",
Move AudioDecoder and related stuff to the api/ directory BUG=webrtc:5805, webrtc:6725 Review-Url: https://codereview.webrtc.org/2668523004 Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 08:15:44 -08:00
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../api/audio_codecs:builtin_audio_encoder_factory",
Move functionality out from AudioFrame and into AudioFrameOperations. This CL is in preparation to move the AudioFrame into webrtc/api. The AudioFrame is a POD type used for representing 10ms of audio. It appears as a parameter and return value of interfaces being migrated to webrtc/api, in particular AudioMixer. Here, methods operator+=, operator>>=, Mute are moved into a new target webrtc/audio/utility/audio_frame_operations, and dependencies are changed to use the new versions. The old AudioFrame methods are marked deprecated. The audio frame utilities in webrtc/modules/utility:audio_frame_operations are also moved to the new location. TBR=kjellander@webrtc.org BUG=webrtc:6548 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2424173003 Cr-Commit-Position: refs/heads/master@{#15413}
2016-12-05 01:46:09 -08:00
"../audio/utility:audio_frame_operations",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
"../call:rtp_interfaces",
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",
Moved RtcEventLog files from call/ to logging/ The RtcEventLog headers need to be accessible from any place which needs logging, and the implementation needs access to data structures that are logged. After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future). The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/). This change allows using RtcEventLog in the p2p/ directory, so that we can log STUN pings and ICE state transitions. BUG=webrtc:6393 R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org Review URL: https://codereview.webrtc.org/2380683005 . Cr-Commit-Position: refs/heads/master@{#14485}
2016-10-03 18:31:22 -07:00
"../logging:rtc_event_log_api",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
"../modules:module_api",
Stop using old AudioCodingModule::RegisterReceiveCodec overloads BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/2388153004 Cr-Commit-Position: refs/heads/master@{#14753}
2016-10-24 13:47:09 -07:00
"../modules/audio_coding:audio_format_conversion",
Cleanup of voice_engine includes. I added a few missing dependencies to the GN target of voice_engine while doing other unrelated work. Currently GN's header include checker has the following to say: $ gn check out/gn_debug webrtc/voice_engine ERROR at //webrtc/voice_engine/include/voe_network.h:38:11: Include not allowed. #include "webrtc/transport.h" ^----------------- It is not in any dependency of //webrtc/voice_engine:voice_engine The include file is in the target(s): //webrtc:webrtc which should somehow be reachable. transport.h should probably move in to webrtc/api, since it is already a pure virtual interface and is used in quite a few places. BUG=webrtc:5589 NOTRY=True Review-Url: https://codereview.webrtc.org/2421483002 Cr-Commit-Position: refs/heads/master@{#14633}
2016-10-13 06:57:16 -07:00
"../modules/audio_coding:rent_a_codec",
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
"../modules/audio_conference_mixer",
"../modules/audio_device",
"../modules/audio_processing",
"../modules/bitrate_controller",
"../modules/media_file",
Prepare the AudioSendStream to be hooked up to send-side BWE. This CL contains three changes as a preparation for adding audio send streams to the send-side BWE: 1. Audio packets are passed through the pacer with high priority. This is needed to be able to set transport sequence numbers on the packets. 2. A feedback observer is passed to the audio stream's rtcp receiver so that the BWE can get notified of any BWE feedback being received on the audio feedback channel. 3. Support for the transport sequence number header extension is added to audio send streams. BUG=webrtc:5263,webrtc:5307 R=mflodman@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1479023002 . Cr-Commit-Position: refs/heads/master@{#10909}
2015-12-07 10:26:18 +01:00
"../modules/pacing",
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
"../modules/rtp_rtcp",
"../modules/utility",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
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
"../system_wrappers",
]
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
}
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
Update formatting of AudioLevel class These changes are all no-op, only affecting the appearance of the code. The file names are changed to match the class name. BUG=none Review-Url: https://codereview.webrtc.org/2731993002 Cr-Commit-Position: refs/heads/master@{#17087}
2017-03-07 01:58:55 -08:00
rtc_static_library("audio_level") {
Broke out 'level_indicator' of the voice_engine GN target. This is needed for the new audio_mixer to depend on just 'voice_engine:level_indicator' instead of the whole voice_engine and avoid a circular dependency. TBR=henrika@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2221103004 Cr-Commit-Position: refs/heads/master@{#13709}
2016-08-10 05:01:51 -07:00
sources = [
Update formatting of AudioLevel class These changes are all no-op, only affecting the appearance of the code. The file names are changed to match the class name. BUG=none Review-Url: https://codereview.webrtc.org/2731993002 Cr-Commit-Position: refs/heads/master@{#17087}
2017-03-07 01:58:55 -08:00
"audio_level.cc",
"audio_level.h",
Broke out 'level_indicator' of the voice_engine GN target. This is needed for the new audio_mixer to depend on just 'voice_engine:level_indicator' instead of the whole voice_engine and avoid a circular dependency. TBR=henrika@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2221103004 Cr-Commit-Position: refs/heads/master@{#13709}
2016-08-10 05:01:51 -07:00
]
deps = [
"..:webrtc_common",
"../common_audio",
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
"../modules:module_api",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base_approved",
Broke out 'level_indicator' of the voice_engine GN target. This is needed for the new audio_mixer to depend on just 'voice_engine:level_indicator' instead of the whole voice_engine and avoid a circular dependency. TBR=henrika@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2221103004 Cr-Commit-Position: refs/heads/master@{#13709}
2016-08-10 05:01:51 -07:00
]
}
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
if (rtc_include_tests) {
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_test("voice_engine_unittests") {
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
deps = [
Enable GN check in voice_engine/ BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2714353002 Cr-Commit-Position: refs/heads/master@{#16986}
2017-03-02 21:28:23 -08:00
":file_player",
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
":voice_engine",
Use relative paths in GN files. BUG=webrtc:7952 TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2974863003 Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 06:20:45 -07:00
"../common_audio",
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
"../modules:module_api",
Use relative paths in GN files. BUG=webrtc:7952 TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2974863003 Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 06:20:45 -07:00
"../modules/audio_coding",
"../modules/audio_conference_mixer",
"../modules/audio_device",
"../modules/audio_processing",
"../modules/media_file",
"../modules/rtp_rtcp",
"../modules/utility",
"../modules/video_capture:video_capture",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
Use relative paths in GN files. BUG=webrtc:7952 TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2974863003 Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 06:20:45 -07:00
"../system_wrappers",
Enable GN check in voice_engine/ BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2714353002 Cr-Commit-Position: refs/heads/master@{#16986}
2017-03-02 21:28:23 -08:00
"../test:test_common",
Use relative paths in GN files. BUG=webrtc:7952 TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2974863003 Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 06:20:45 -07:00
"../test:test_main",
"../test:video_test_common",
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
"//testing/gmock",
"//testing/gtest",
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
"//third_party/gflags",
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
]
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
GN: Update tests to have the correct shard timeout value on Android. TBR=mflodman@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2231413002 Cr-Commit-Position: refs/heads/master@{#13741}
2016-08-15 02:29:11 -07:00
shard_timeout = 900
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
}
sources = [
"channel_unittest.cc",
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
"file_player_unittests.cc",
Add TransportFeedbackPacketLossTracker. This CL is to calculate packet loss metrics from TransportFeedback. The outcome of this will be passed down to audio encoder. BUG=webrtc:6904 Review-Url: https://codereview.webrtc.org/2579613003 Cr-Commit-Position: refs/heads/master@{#16217}
2017-01-23 08:07:05 -08:00
"transport_feedback_packet_loss_tracker_unittest.cc",
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
"utility_unittest.cc",
"voe_base_unittest.cc",
"voe_codec_unittest.cc",
"voe_network_unittest.cc",
"voice_engine_fixture.cc",
"voice_engine_fixture.h",
]
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
data = [
Use relative paths in GN files. BUG=webrtc:7952 TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2974863003 Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 06:20:45 -07:00
"../../resources/utility/encapsulated_pcm16b_8khz.wav",
"../../resources/utility/encapsulated_pcmu_8khz.wav",
Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978}
2017-01-10 01:12:51 -08:00
]
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
if (is_win) {
defines = [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]
cflags = [
# TODO(kjellander): Bug 261: fix this warning.
"/wd4373", # Virtual function override.
]
}
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
}
}
if (!is_ios) {
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("voe_auto_test") {
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
testonly = true
deps = [
":voice_engine",
Enable GN check in voice_engine/ BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2714353002 Cr-Commit-Position: refs/heads/master@{#16986}
2017-03-02 21:28:23 -08:00
"..:webrtc_common",
Use relative paths in GN files. BUG=webrtc:7952 TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2974863003 Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 06:20:45 -07:00
"../logging:rtc_event_log_api",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"../modules:module_api",
Enable GN check in voice_engine/ BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2714353002 Cr-Commit-Position: refs/heads/master@{#16986}
2017-03-02 21:28:23 -08:00
"../modules/audio_device:audio_device",
"../modules/audio_processing:audio_processing",
"../modules/rtp_rtcp:rtp_rtcp",
Use relative paths in GN files. BUG=webrtc:7952 TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2974863003 Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 06:20:45 -07:00
"../modules/video_capture",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base_approved",
Use relative paths in GN files. BUG=webrtc:7952 TBR=kjellander@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2974863003 Cr-Commit-Position: refs/heads/master@{#18970}
2017-07-11 06:20:45 -07:00
"../system_wrappers",
"../system_wrappers/:system_wrappers_default",
"../test/:test_common",
"../test/:test_support",
"../test/:video_test_common",
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
sources = [
"test/auto_test/automated_mode.cc",
"test/auto_test/fakes/conference_transport.cc",
"test/auto_test/fakes/conference_transport.h",
"test/auto_test/fakes/loudest_filter.cc",
"test/auto_test/fakes/loudest_filter.h",
"test/auto_test/fixtures/after_initialization_fixture.cc",
"test/auto_test/fixtures/after_initialization_fixture.h",
"test/auto_test/fixtures/after_streaming_fixture.cc",
"test/auto_test/fixtures/after_streaming_fixture.h",
"test/auto_test/fixtures/before_initialization_fixture.cc",
"test/auto_test/fixtures/before_initialization_fixture.h",
"test/auto_test/fixtures/before_streaming_fixture.cc",
"test/auto_test/fixtures/before_streaming_fixture.h",
"test/auto_test/standard/codec_before_streaming_test.cc",
"test/auto_test/standard/codec_test.cc",
"test/auto_test/standard/dtmf_test.cc",
"test/auto_test/standard/rtp_rtcp_before_streaming_test.cc",
"test/auto_test/standard/rtp_rtcp_extensions.cc",
"test/auto_test/standard/rtp_rtcp_test.cc",
"test/auto_test/voe_conference_test.cc",
"test/auto_test/voe_standard_test.cc",
"test/auto_test/voe_standard_test.h",
"test/auto_test/voe_test_defines.h",
]
defines = []
if (rtc_enable_protobuf) {
defines = [ "ENABLE_RTC_EVENT_LOG" ]
}
if (is_win) {
defines += [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]
cflags = [
"/wd4267", # size_t to int truncation.
"/wd4373", # Virtual function override.
# TODO(kjellander): Bug 261: fix this warning.
]
}
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
Added voice_engine_unittests and voe_auto_test targets to GN. I didn't convert the APK tests, since I couldn't figure it out, nor find any that had already been converted to copy from. I might get to it in a new CL or even second patch-set. BUG=webrtc:6039 Review-Url: https://codereview.webrtc.org/2123243003 Cr-Commit-Position: refs/heads/master@{#13410}
2016-07-08 02:12:55 -07:00
}
}
}
}
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