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webrtc_m130/webrtc/modules/audio_processing/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.
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
import("//build/config/arm.gni")
import("//third_party/protobuf/proto_library.gni")
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
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
declare_args() {
# Disables the usual mode where we trust the reported system delay
# values the AEC receives. The corresponding define is set appropriately
# in the code, but it can be force-enabled here for testing.
aec_untrusted_delay_for_testing = false
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("audio_processing") {
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 = [
Moved the AEC C code to be built using C++. In order for the change to be reviewable, the move was made into two steps consisting of the first two patches in this CL. Step 1 (patch set 1): -Changed file types to use .cc -Changed buildfiles to use the new files -Changed C code inclusion to properly match the changed file formats (removed and added extern "C" declarations). -Changed implicit void-> nonvoid casts that are illegal in C++ to be explicit. Step 2 (patch set 2): -Changed all the warnings reported when uploading the CL. -The warnings about formatting of the assembly optimized code were not addressed though. BUG=webrtc:5201 Review URL: https://codereview.webrtc.org/1713923002 Cr-Commit-Position: refs/heads/master@{#11727}
2016-02-23 14:34:59 -08:00
"aec/aec_core.cc",
Merge audio_processing changes. R=aluebs@webrtc.org, bjornv@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/32769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:41:24 +00:00
"aec/aec_core.h",
Moved the functionality in aec_core_internal.h into other files. The purpose of this CL is to simplify upcoming AEC algorithm changes. The changes should be bitexact. The presubmit was bypassed due to a presubmit complaint about usage of short instead of int16_t which will be addressed in upcoming CLs. BUG=webrtc:5298, webrtc:5201 NOPRESUBMIT=true Review-Url: https://codereview.webrtc.org/1949803004 Cr-Commit-Position: refs/heads/master@{#12662}
2016-05-09 03:57:33 -07:00
"aec/aec_core_optimized_methods.h",
Moved the file aec_resampler.c to be built using C++. The steps involved were: 1) Change file name to .cc from .c. 2) Update the build files accordingly. 3) Remove the extern header file inclusion. 4) Change the casts in aec_resampler.cc to static_cast and reinterpret_cast. The changes are bitexact. The CL will be followed with another CL where a proper (webrtc) namespace is introduced. The reason for not having it in this CL is that this was missed in the corresponding CL that did the above for aec_core.c, ..., and if the namespaces in all the aec_core -related files can be changed at the same time that will simplify things. BUG=webrtc:5201 Review URL: https://codereview.webrtc.org/1754223004 Cr-Commit-Position: refs/heads/master@{#11867}
2016-03-04 00:12:40 -08:00
"aec/aec_resampler.cc",
Merge audio_processing changes. R=aluebs@webrtc.org, bjornv@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/32769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:41:24 +00:00
"aec/aec_resampler.h",
Moved the AEC C code to be built using C++. In order for the change to be reviewable, the move was made into two steps consisting of the first two patches in this CL. Step 1 (patch set 1): -Changed file types to use .cc -Changed buildfiles to use the new files -Changed C code inclusion to properly match the changed file formats (removed and added extern "C" declarations). -Changed implicit void-> nonvoid casts that are illegal in C++ to be explicit. Step 2 (patch set 2): -Changed all the warnings reported when uploading the CL. -The warnings about formatting of the assembly optimized code were not addressed though. BUG=webrtc:5201 Review URL: https://codereview.webrtc.org/1713923002 Cr-Commit-Position: refs/heads/master@{#11727}
2016-02-23 14:34:59 -08:00
"aec/echo_cancellation.cc",
Remove webrtc/modules/audio_processing/{aec,aecm,ns}/include 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=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1440523002 . Cr-Commit-Position: refs/heads/master@{#10608}
2015-11-11 20:16:11 +01:00
"aec/echo_cancellation.h",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/adaptive_fir_filter.cc",
"aec3/adaptive_fir_filter.h",
"aec3/aec3_common.cc",
"aec3/aec3_common.h",
"aec3/aec3_fft.cc",
"aec3/aec3_fft.h",
"aec3/aec_state.cc",
"aec3/aec_state.h",
Reland of Added first layer of the echo canceller 3 functionality. Reason for reland: Added disabling of the BlockProcessor DEATH tests due to false alarms on the trybots. These errors cannot be reproduced locally and they occur only in the DEATH test code, which means that the memory leaks do not occur under normal conditions. Original issue's description: > Reason for revert: > Memcheck buildbot detected memory leaks in the death tests. The code looks fine > to me, but please investigate what causes the error and reland. > > Original issue's description: > > Added first layer of the echo canceller 3 functionality. > > > > This CL adds the first layer of the echo canceller 3. > > All of the code is as it should, apart from > > block_processor.* which only contains placeholder > > functionality. (Upcoming CLs will add proper > > functionality into those files.) > > > > > > > > BUG=webrtc:6018 > > > > Review-Url: https://codereview.webrtc.org/2584493002 > > Cr-Commit-Position: refs/heads/master@{#15861} > > Committed: > https://chromium.googlesource.com/external/webrtc/+/38fd1758e90bcdc7690a552e7ef0ec0d143d2f30 > > TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org,peah@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6018 BUG=webrtc:6018 TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org Review-Url: https://codereview.webrtc.org/2608233002 Cr-Commit-Position: refs/heads/master@{#15884}
2017-01-03 04:20:34 -08:00
"aec3/block_framer.cc",
"aec3/block_framer.h",
"aec3/block_processor.cc",
"aec3/block_processor.h",
Adding metrics to AEC3. This CL adds metrics reporting to AEC3. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2722453002 Cr-Commit-Position: refs/heads/master@{#16929}
2017-02-28 22:08:53 -08:00
"aec3/block_processor_metrics.cc",
"aec3/block_processor_metrics.h",
Reland of Added first layer of the echo canceller 3 functionality. Reason for reland: Added disabling of the BlockProcessor DEATH tests due to false alarms on the trybots. These errors cannot be reproduced locally and they occur only in the DEATH test code, which means that the memory leaks do not occur under normal conditions. Original issue's description: > Reason for revert: > Memcheck buildbot detected memory leaks in the death tests. The code looks fine > to me, but please investigate what causes the error and reland. > > Original issue's description: > > Added first layer of the echo canceller 3 functionality. > > > > This CL adds the first layer of the echo canceller 3. > > All of the code is as it should, apart from > > block_processor.* which only contains placeholder > > functionality. (Upcoming CLs will add proper > > functionality into those files.) > > > > > > > > BUG=webrtc:6018 > > > > Review-Url: https://codereview.webrtc.org/2584493002 > > Cr-Commit-Position: refs/heads/master@{#15861} > > Committed: > https://chromium.googlesource.com/external/webrtc/+/38fd1758e90bcdc7690a552e7ef0ec0d143d2f30 > > TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org,peah@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6018 BUG=webrtc:6018 TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org Review-Url: https://codereview.webrtc.org/2608233002 Cr-Commit-Position: refs/heads/master@{#15884}
2017-01-03 04:20:34 -08:00
"aec3/cascaded_biquad_filter.cc",
"aec3/cascaded_biquad_filter.h",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/comfort_noise_generator.cc",
"aec3/comfort_noise_generator.h",
Adding full initial version of delay estimation functionality in echo canceller 3 This CL adds code to the all the delay estimation functionality that is available for the first version of echo canceller 3. The code completes the class EchoPathDelayEstimator. Note that this code does not yet include any handling of clock-drift so there will be upcoming versions of this code. Also note that the CL includes some minor changes in other files for echo canceller 3. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2644123002 Cr-Commit-Position: refs/heads/master@{#16489}
2017-02-08 05:08:56 -08:00
"aec3/decimator_by_4.cc",
"aec3/decimator_by_4.h",
This CL adds the basic framework for AEC3 in the audio processing module. It will be followed by a number of other CLs that extends this framework. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2567513003 Cr-Commit-Position: refs/heads/master@{#15593}
2016-12-14 01:16:23 -08:00
"aec3/echo_canceller3.cc",
"aec3/echo_canceller3.h",
Adding second layer of the echo canceller 3 functionality. This CL adds code to the BlockProcessor, which basically constitutes the second layer in echo canceller 3. The CL includes two incomplete classes (EchoRemover and EchoPathDelayEstimator) which will be completed in upcoming CLs. Because of this, some of the unittests are disabled until those are added. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2611223003 Cr-Commit-Position: refs/heads/master@{#16319}
2017-01-27 03:28:19 -08:00
"aec3/echo_path_delay_estimator.cc",
"aec3/echo_path_delay_estimator.h",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/echo_path_variability.cc",
Adding second layer of the echo canceller 3 functionality. This CL adds code to the BlockProcessor, which basically constitutes the second layer in echo canceller 3. The CL includes two incomplete classes (EchoRemover and EchoPathDelayEstimator) which will be completed in upcoming CLs. Because of this, some of the unittests are disabled until those are added. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2611223003 Cr-Commit-Position: refs/heads/master@{#16319}
2017-01-27 03:28:19 -08:00
"aec3/echo_path_variability.h",
"aec3/echo_remover.cc",
"aec3/echo_remover.h",
Adding metrics to AEC3. This CL adds metrics reporting to AEC3. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2722453002 Cr-Commit-Position: refs/heads/master@{#16929}
2017-02-28 22:08:53 -08:00
"aec3/echo_remover_metrics.cc",
"aec3/echo_remover_metrics.h",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/erl_estimator.cc",
"aec3/erl_estimator.h",
"aec3/erle_estimator.cc",
"aec3/erle_estimator.h",
"aec3/fft_buffer.cc",
"aec3/fft_buffer.h",
"aec3/fft_data.h",
Reland of Added first layer of the echo canceller 3 functionality. Reason for reland: Added disabling of the BlockProcessor DEATH tests due to false alarms on the trybots. These errors cannot be reproduced locally and they occur only in the DEATH test code, which means that the memory leaks do not occur under normal conditions. Original issue's description: > Reason for revert: > Memcheck buildbot detected memory leaks in the death tests. The code looks fine > to me, but please investigate what causes the error and reland. > > Original issue's description: > > Added first layer of the echo canceller 3 functionality. > > > > This CL adds the first layer of the echo canceller 3. > > All of the code is as it should, apart from > > block_processor.* which only contains placeholder > > functionality. (Upcoming CLs will add proper > > functionality into those files.) > > > > > > > > BUG=webrtc:6018 > > > > Review-Url: https://codereview.webrtc.org/2584493002 > > Cr-Commit-Position: refs/heads/master@{#15861} > > Committed: > https://chromium.googlesource.com/external/webrtc/+/38fd1758e90bcdc7690a552e7ef0ec0d143d2f30 > > TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org,peah@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6018 BUG=webrtc:6018 TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org Review-Url: https://codereview.webrtc.org/2608233002 Cr-Commit-Position: refs/heads/master@{#15884}
2017-01-03 04:20:34 -08:00
"aec3/frame_blocker.cc",
"aec3/frame_blocker.h",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/main_filter_update_gain.cc",
"aec3/main_filter_update_gain.h",
Adding full initial version of delay estimation functionality in echo canceller 3 This CL adds code to the all the delay estimation functionality that is available for the first version of echo canceller 3. The code completes the class EchoPathDelayEstimator. Note that this code does not yet include any handling of clock-drift so there will be upcoming versions of this code. Also note that the CL includes some minor changes in other files for echo canceller 3. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2644123002 Cr-Commit-Position: refs/heads/master@{#16489}
2017-02-08 05:08:56 -08:00
"aec3/matched_filter.cc",
"aec3/matched_filter.h",
"aec3/matched_filter_lag_aggregator.cc",
"aec3/matched_filter_lag_aggregator.h",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/output_selector.cc",
"aec3/output_selector.h",
"aec3/power_echo_model.cc",
"aec3/power_echo_model.h",
Adding second layer of the echo canceller 3 functionality. This CL adds code to the BlockProcessor, which basically constitutes the second layer in echo canceller 3. The CL includes two incomplete classes (EchoRemover and EchoPathDelayEstimator) which will be completed in upcoming CLs. Because of this, some of the unittests are disabled until those are added. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2611223003 Cr-Commit-Position: refs/heads/master@{#16319}
2017-01-27 03:28:19 -08:00
"aec3/render_delay_buffer.cc",
"aec3/render_delay_buffer.h",
"aec3/render_delay_controller.cc",
"aec3/render_delay_controller.h",
Adding metrics to AEC3. This CL adds metrics reporting to AEC3. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2722453002 Cr-Commit-Position: refs/heads/master@{#16929}
2017-02-28 22:08:53 -08:00
"aec3/render_delay_controller_metrics.cc",
"aec3/render_delay_controller_metrics.h",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/render_signal_analyzer.cc",
"aec3/render_signal_analyzer.h",
"aec3/residual_echo_estimator.cc",
"aec3/residual_echo_estimator.h",
"aec3/shadow_filter_update_gain.cc",
"aec3/shadow_filter_update_gain.h",
"aec3/subtractor.cc",
"aec3/subtractor.h",
"aec3/subtractor_output.h",
"aec3/suppression_filter.cc",
"aec3/suppression_filter.h",
"aec3/suppression_gain.cc",
"aec3/suppression_gain.h",
What was done was -changed filenames to *.cc -fixed issues with implicit casts causing build errors. All other CL warnings were ignored as the original code is not changed, merely moved. BUG=webrtc:5724 NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1857153002 Cr-Commit-Position: refs/heads/master@{#12307}
2016-04-10 22:38:14 -07:00
"aecm/aecm_core.cc",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"aecm/aecm_core.h",
What was done was -changed filenames to *.cc -fixed issues with implicit casts causing build errors. All other CL warnings were ignored as the original code is not changed, merely moved. BUG=webrtc:5724 NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1857153002 Cr-Commit-Position: refs/heads/master@{#12307}
2016-04-10 22:38:14 -07:00
"aecm/echo_control_mobile.cc",
Remove webrtc/modules/audio_processing/{aec,aecm,ns}/include 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=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1440523002 . Cr-Commit-Position: refs/heads/master@{#10608}
2015-11-11 20:16:11 +01:00
"aecm/echo_control_mobile.h",
Merge audio_processing changes. R=aluebs@webrtc.org, bjornv@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/32769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:41:24 +00:00
"agc/agc.cc",
"agc/agc.h",
"agc/agc_manager_direct.cc",
"agc/agc_manager_direct.h",
"agc/gain_map_internal.h",
Change name of files and class in agc/histogram* in order to avoid issue file-name clash in build files The changes are done in several patches in order to make the review easier. NOTRY=True BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2051443002 Cr-Commit-Position: refs/heads/master@{#13068}
2016-06-08 06:42:02 -07:00
"agc/loudness_histogram.cc",
"agc/loudness_histogram.h",
Merge audio_processing changes. R=aluebs@webrtc.org, bjornv@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/32769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:41:24 +00:00
"agc/utility.cc",
"agc/utility.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
"audio_buffer.cc",
"audio_buffer.h",
"audio_processing_impl.cc",
"audio_processing_impl.h",
Make the separation between target and interferer scenario depend on microphone spacing in NonlinearBeamformer Depends on this CL: https://codereview.webrtc.org/1378973003/ Review URL: https://codereview.webrtc.org/1388033002 Cr-Commit-Position: refs/heads/master@{#10330}
2015-10-19 18:02:39 -07:00
"beamformer/array_util.cc",
"beamformer/array_util.h",
Merge beamformer R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7958 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 22:22:04 +00:00
"beamformer/complex_matrix.h",
"beamformer/covariance_matrix_generator.cc",
"beamformer/covariance_matrix_generator.h",
"beamformer/matrix.h",
Remove build-time beamformer flags. RealFourier is now unconditionally enabled since we can fall back to the Ooura FFT. We no longer need to condition users on rtc_use_openmax_dl. R=aluebs@webrtc.org, mgraczyk@google.com Review URL: https://webrtc-codereview.appspot.com/50439004 Cr-Commit-Position: refs/heads/master@{#8799} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8799 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 00:28:22 +00:00
"beamformer/nonlinear_beamformer.cc",
"beamformer/nonlinear_beamformer.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
"common.h",
"echo_cancellation_impl.cc",
"echo_cancellation_impl.h",
"echo_control_mobile_impl.cc",
"echo_control_mobile_impl.h",
Add algorithm for Residual Echo Detector. This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values. BUG=webrtc:6525 Review-Url: https://codereview.webrtc.org/2419563003 Cr-Commit-Position: refs/heads/master@{#14824}
2016-10-28 07:04:03 -07:00
"echo_detector/circular_buffer.cc",
"echo_detector/circular_buffer.h",
"echo_detector/mean_variance_estimator.cc",
"echo_detector/mean_variance_estimator.h",
Added a new echo likelihood stat that reports the maximum value from a previous time period. BUG=webrtc:6797 Review-Url: https://codereview.webrtc.org/2629563003 Cr-Commit-Position: refs/heads/master@{#16079}
2017-01-15 08:29:46 -08:00
"echo_detector/moving_max.cc",
"echo_detector/moving_max.h",
Add algorithm for Residual Echo Detector. This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values. BUG=webrtc:6525 Review-Url: https://codereview.webrtc.org/2419563003 Cr-Commit-Position: refs/heads/master@{#14824}
2016-10-28 07:04:03 -07:00
"echo_detector/normalized_covariance_estimator.cc",
"echo_detector/normalized_covariance_estimator.h",
Moved the GainControlForNewAGC class to be a separate file. Apart from being motivated in order to make the source files shorter this is needed when separating the APM submodules structs into separate files. BUG= Review URL: https://codereview.webrtc.org/1678813002 Cr-Commit-Position: refs/heads/master@{#11611}
2016-02-13 16:40:47 -08:00
"gain_control_for_experimental_agc.cc",
"gain_control_for_experimental_agc.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
"gain_control_impl.cc",
"gain_control_impl.h",
This CL adds functionality in the level controller to receive a signal level to use initially, instead of the default initial signal level. The initial form of the CL (https://codereview.webrtc.org/2254973003/) was reverted due to down-stream dependencies. These have been resolved, but the CL needed to be revised according to the new scheme for passing parameters to the audio processing module. Therefore, please review this CL as if it is new. TBR=aleloi@webrtc.org BUG=webrtc:6386 Review-Url: https://codereview.webrtc.org/2337083002 Cr-Commit-Position: refs/heads/master@{#14579}
2016-10-07 14:54:10 -07:00
"include/audio_processing.cc",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"include/audio_processing.h",
Moving/renaming webrtc/common.h. This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code. - For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel(). - Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state. BUG=webrtc:5879 Review-Url: https://codereview.webrtc.org/2307533004 Cr-Commit-Position: refs/heads/master@{#14109}
2016-09-07 07:34:41 -07:00
"include/config.cc",
"include/config.h",
New module for the adaptive level controlling functionality in the audio processing module NOTRY=true TBR=aluebs@webrtc.org BUG=webrtc:5920 Review-Url: https://codereview.webrtc.org/2090583002 Cr-Commit-Position: refs/heads/master@{#13333}
2016-06-29 15:26:12 -07:00
"level_controller/biquad_filter.cc",
"level_controller/biquad_filter.h",
"level_controller/down_sampler.cc",
"level_controller/down_sampler.h",
"level_controller/gain_applier.cc",
"level_controller/gain_applier.h",
"level_controller/gain_selector.cc",
"level_controller/gain_selector.h",
"level_controller/level_controller.cc",
"level_controller/level_controller.h",
This CL adds functionality in the level controller to receive a signal level to use initially, instead of the default initial signal level. The initial form of the CL (https://codereview.webrtc.org/2254973003/) was reverted due to down-stream dependencies. These have been resolved, but the CL needed to be revised according to the new scheme for passing parameters to the audio processing module. Therefore, please review this CL as if it is new. TBR=aleloi@webrtc.org BUG=webrtc:6386 Review-Url: https://codereview.webrtc.org/2337083002 Cr-Commit-Position: refs/heads/master@{#14579}
2016-10-07 14:54:10 -07:00
"level_controller/level_controller_constants.h",
New module for the adaptive level controlling functionality in the audio processing module NOTRY=true TBR=aluebs@webrtc.org BUG=webrtc:5920 Review-Url: https://codereview.webrtc.org/2090583002 Cr-Commit-Position: refs/heads/master@{#13333}
2016-06-29 15:26:12 -07:00
"level_controller/noise_level_estimator.cc",
"level_controller/noise_level_estimator.h",
"level_controller/noise_spectrum_estimator.cc",
"level_controller/noise_spectrum_estimator.h",
"level_controller/peak_level_estimator.cc",
"level_controller/peak_level_estimator.h",
"level_controller/saturating_gain_estimator.cc",
"level_controller/saturating_gain_estimator.h",
"level_controller/signal_classifier.cc",
"level_controller/signal_classifier.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
"level_estimator_impl.cc",
"level_estimator_impl.h",
This CL introduces a new data logging functionality to use for the APM. It allows simple and rapid additions of exploratory data logpoints to use during bug investigations and module performance analysis. The new data logging functionality is also in this CL used to replace the existing data logging functionality present in the AEC. Additional information: As there was an issue with that the build flag for activating this feature was not present in all compilation units that included the feature additional changes were needed. A summary of the changes are -The build files were modified to ensure that the logging build flag always is set to either 0 or 1 for compilation units that include the feature. -Build-time checks in the appropriate places were added to ensure that the above is fulfilled. -The build object was added dynamically to the AEC state as a pointer to ensure that the size of that state is not dependent on whether the logging build flag is set or not. -The constructor of the AEC class needed to be modified in order to construct the logging object. For this a destructor was also needed. -An unused method without any declaration was removed in order to avoid any issues with the logging flag being set to 0 or 1. This CL will be immediately followed with an upcoming CL that replaces the logging in echo_cancellation.cc with the new functionality which will ensure that the logging flag is only used in one place within WebRTC, which in turn will fully ensure that all compilation units that uses the feature also have the flag properly set. BUG=webrtc:5201, webrtc:5298 Review-Url: https://codereview.webrtc.org/1877713002 Cr-Commit-Position: refs/heads/master@{#12607}
2016-05-03 07:01:18 -07:00
"logging/apm_data_dumper.cc",
"logging/apm_data_dumper.h",
This CL introduces the new functionality for setting the APM parameters to the high-pass filter. The introduction will be done in three steps: 1) This CL which introduces the new scheme and changes the code in webrtcvoiceengine.cc to use it. 2) Introduce the scheme into upstream code. 3) Remove the HighPassFilter interface in APM. BUG=webrtc::6220, webrtc::6296, webrtc::6297, webrtc::6181, webrtc::5298 Review-Url: https://codereview.webrtc.org/2415403002 Cr-Commit-Position: refs/heads/master@{#15197}
2016-11-22 07:24:52 -08:00
"low_cut_filter.cc",
"low_cut_filter.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
"noise_suppression_impl.cc",
"noise_suppression_impl.h",
Removed the ProcessingComponent class BUG= Review URL: https://codereview.webrtc.org/1772553002 Cr-Commit-Position: refs/heads/master@{#11950}
2016-03-10 23:05:28 -08:00
"render_queue_item_verifier.h",
Add empty residual echo detector. This CL does not contain the actual algorithm, but only creates an empty processing component and connects the right signals to it. The algorithm will be added in a follow-up CL. BUG=webrtc:6525 Review-Url: https://codereview.webrtc.org/2405403003 Cr-Commit-Position: refs/heads/master@{#14820}
2016-10-28 05:39:16 -07:00
"residual_echo_detector.cc",
"residual_echo_detector.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
"rms_level.cc",
"rms_level.h",
Wrap the splitting filter in its own class This doesn't change the behavior at all. The logic behind this is having one class which manages all the splitting filters, because in the future we plan to add a 3 band one for 48kHz support. It also breaks the dependency of the AudioBuffer with the filter states of these filters (which are going to be different for the 3 band one). The AudioBuffer is complicated enough and is going to need changes to support 3 bands in the future, so any simplification is a good idea. On top of that it eliminates repeated code in the APM (now only iterating over channels, but then also deciding in how many bands to split). This should be managed by the AudioBuffer directly. BUG=webrtc:3146 R=bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7705 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 22:18:10 +00:00
"splitting_filter.cc",
"splitting_filter.h",
Add 3-band filter-bank implementation The implementation is a FIR filter bank with DCT modulation, similar to the proposed in "Multirate Signal Processing for Communication Systems" by Fredric J Harris. The lowpass filter prototype has these characteristics: * Passband ripple = 0.3dB * Passband frequency = 0.147 (7kHz at 48kHz) * Stopband attenuation = 40dB * Stopband frequency = 0.192 (9.2kHz at 48kHz) * Delay = 24 samples (500us at 48kHz) * Linear phase This filter bank does not satisfy perfect reconstruction. The SNR after analysis and synthesis (with no processing in between) is approximately 9.5dB depending on the input signal after compensating for the delay. The performance on my workstation of AudioProcessing (with AGC and NS enabled) on a 413s recording compared to previous versions is as follows: * Input signal has 32kHz sample rate: 3.01s * Resampling 48kHz to 32kHz: 3.56s * Today's temporary filter bank: 5.67s * This filter-bank: 4.62s BUG=webrtc:3146 R=andrew@webrtc.org, bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48999005 Cr-Commit-Position: refs/heads/master@{#9090}
2015-04-27 11:34:45 -07:00
"three_band_filter_bank.cc",
"three_band_filter_bank.h",
Merge audio_processing changes. R=aluebs@webrtc.org, bjornv@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/32769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:41:24 +00:00
"transient/common.h",
"transient/daubechies_8_wavelet_coeffs.h",
"transient/dyadic_decimator.h",
"transient/moving_moments.cc",
"transient/moving_moments.h",
"transient/transient_detector.cc",
"transient/transient_detector.h",
"transient/transient_suppressor.cc",
"transient/transient_suppressor.h",
"transient/wpd_node.cc",
"transient/wpd_node.h",
"transient/wpd_tree.cc",
"transient/wpd_tree.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
"typing_detection.cc",
"typing_detection.h",
Adding BlockMeanCalculator for AEC. This will improve the readability of AEC code. BUG= Review URL: https://codereview.webrtc.org/1805633006 Cr-Commit-Position: refs/heads/master@{#12123}
2016-03-24 14:36:25 -07:00
"utility/block_mean_calculator.cc",
"utility/block_mean_calculator.h",
Changed the delay estimator to be built using C++ BUG=webrtc:5724 NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1878613002 Cr-Commit-Position: refs/heads/master@{#12336}
2016-04-12 14:47:40 -07:00
"utility/delay_estimator.cc",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"utility/delay_estimator.h",
"utility/delay_estimator_internal.h",
Changed the delay estimator to be built using C++ BUG=webrtc:5724 NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1878613002 Cr-Commit-Position: refs/heads/master@{#12336}
2016-04-12 14:47:40 -07:00
"utility/delay_estimator_wrapper.cc",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"utility/delay_estimator_wrapper.h",
The FFT functionality in aec_rdft* is based on legacy C code which is not thread-safe in the sense that the rdft_init method can only be run in a single-threaded. Currently, inside WebRTC multiple instances of the audio- processing module are set up which means that the init method may be run concurrently. In order to avoid having to protect the init method with a lock to ensure single-threaded behavior that, this CL places the FFT functionality inside a class so that there is no global component of the FFT functionality. Note that: 1) The nonstandard header for the ooura_fft.cc was copied from the aec_rdft.cc header, and augmented with a description of the changes introduced in this CL. 2) The clang warnings for the ooura_fft_sse2.cc, ooura_fft_neon.cc and ooura_fft_mips.cc were not addressed as this code was kept as it was before this CL 3) Clang-format was run on all files apart from ooura_fft_mips.cc (as that would change the format of the inline assempbly code). Adding bypass of presubmit to avoid code style and header errors caused by the fact that files with legacy code are being renamed. NOPRESUBMIT=true BUG=chromium:638583 Review-Url: https://codereview.webrtc.org/2348213002 Cr-Commit-Position: refs/heads/master@{#14554}
2016-10-06 06:46:20 -07:00
"utility/ooura_fft.cc",
"utility/ooura_fft.h",
"utility/ooura_fft_tables_common.h",
Pull the Voice Activity Detector out from the AGC This change generates bit-exact values when running through audioproc_f than before. This change was originally uploaded here: * https://codereview.webrtc.org/1181933002/ * https://codereview.webrtc.org/1177043017/ And reverted because of an ASAN problem in Chrome here: * https://codereview.webrtc.org/1192863006/ * https://codereview.webrtc.org/1194963003/ TBR=andrew@webrtc.org Review URL: https://codereview.webrtc.org/1212543002 Cr-Commit-Position: refs/heads/master@{#9505}
2015-06-25 12:28:48 -07:00
"vad/common.h",
"vad/gmm.cc",
"vad/gmm.h",
"vad/noise_gmm_tables.h",
"vad/pitch_based_vad.cc",
"vad/pitch_based_vad.h",
"vad/pitch_internal.cc",
"vad/pitch_internal.h",
"vad/pole_zero_filter.cc",
"vad/pole_zero_filter.h",
"vad/standalone_vad.cc",
"vad/standalone_vad.h",
"vad/vad_audio_proc.cc",
"vad/vad_audio_proc.h",
"vad/vad_audio_proc_internal.h",
"vad/vad_circular_buffer.cc",
"vad/vad_circular_buffer.h",
"vad/voice_activity_detector.cc",
"vad/voice_activity_detector.h",
"vad/voice_gmm_tables.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
"voice_detection_impl.cc",
"voice_detection_impl.h",
]
defines = []
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
"../..:webrtc_common",
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",
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
"../../base:gtest_prod",
Don't link with audio codecs that we don't use We used to link with all audio codecs unconditionally (except Opus); this patch makes gyp and gn only link to the ones that are used. This unfortunately fails to have a measurable impact on Chromium binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC fix were already being excluded from Chromium by some other means, likely just the linker omitting compilation units with no incoming references. (This was previously landed as revisions 10046 and 10060, and got reverted because it broke several of the Chromium FYI bots.) BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1368843003 Cr-Commit-Position: refs/heads/master@{#10127}
2015-09-30 21:54:21 -07:00
"../audio_coding:isac",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
public_deps = [
":audio_processing_c",
]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
Moved the place for the aec_debug_dump build flag and changed the name to apm_debug_dump Currently, the aec_debug_dump buildflag can and is used to store data in the whole of the audio processing module. Therefore a more appropriate name is apm_debug_dump which also matches the names of the data dumping functionality. This CL makes that name change. The CL also changes the WEBRTC_AEC_DEBUG_DUMP define to WEBRTC_APM_DEBUG_DUMP == 1 Furthermore, this CL moves the buildflag to a more appropriate place. BUG=webrtc:5298 Review-Url: https://codereview.webrtc.org/2300813004 Cr-Commit-Position: refs/heads/master@{#14026}
2016-09-01 08:58:21 -07:00
if (apm_debug_dump) {
defines += [ "WEBRTC_APM_DEBUG_DUMP=1" ]
This CL introduces a new data logging functionality to use for the APM. It allows simple and rapid additions of exploratory data logpoints to use during bug investigations and module performance analysis. The new data logging functionality is also in this CL used to replace the existing data logging functionality present in the AEC. Additional information: As there was an issue with that the build flag for activating this feature was not present in all compilation units that included the feature additional changes were needed. A summary of the changes are -The build files were modified to ensure that the logging build flag always is set to either 0 or 1 for compilation units that include the feature. -Build-time checks in the appropriate places were added to ensure that the above is fulfilled. -The build object was added dynamically to the AEC state as a pointer to ensure that the size of that state is not dependent on whether the logging build flag is set or not. -The constructor of the AEC class needed to be modified in order to construct the logging object. For this a destructor was also needed. -An unused method without any declaration was removed in order to avoid any issues with the logging flag being set to 0 or 1. This CL will be immediately followed with an upcoming CL that replaces the logging in echo_cancellation.cc with the new functionality which will ensure that the logging flag is only used in one place within WebRTC, which in turn will fully ensure that all compilation units that uses the feature also have the flag properly set. BUG=webrtc:5201, webrtc:5298 Review-Url: https://codereview.webrtc.org/1877713002 Cr-Commit-Position: refs/heads/master@{#12607}
2016-05-03 07:01:18 -07:00
} else {
Moved the place for the aec_debug_dump build flag and changed the name to apm_debug_dump Currently, the aec_debug_dump buildflag can and is used to store data in the whole of the audio processing module. Therefore a more appropriate name is apm_debug_dump which also matches the names of the data dumping functionality. This CL makes that name change. The CL also changes the WEBRTC_AEC_DEBUG_DUMP define to WEBRTC_APM_DEBUG_DUMP == 1 Furthermore, this CL moves the buildflag to a more appropriate place. BUG=webrtc:5298 Review-Url: https://codereview.webrtc.org/2300813004 Cr-Commit-Position: refs/heads/master@{#14026}
2016-09-01 08:58:21 -07:00
defines += [ "WEBRTC_APM_DEBUG_DUMP=0" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
if (aec_untrusted_delay_for_testing) {
defines += [ "WEBRTC_UNTRUSTED_DELAY" ]
}
GN: Prefix WebRTC specific variables with "rtc_" BUG=3441 TESTED=Trybots + Running GN in a Chromium checkout with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using: --args="os=\"android\" cpu_arch=\"arm\"" R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/27379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:36:10 +00:00
if (rtc_enable_protobuf) {
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 += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
deps += [ ":audioproc_debug_proto" ]
}
Deactivated the intelligibility enhancement functionality by default NOTRY=true BUG= Review-Url: https://codereview.webrtc.org/2272423003 Cr-Commit-Position: refs/heads/master@{#13937}
2016-08-26 07:16:04 -07:00
if (rtc_enable_intelligibility_enhancer) {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
sources += [
"intelligibility/intelligibility_enhancer.cc",
"intelligibility/intelligibility_enhancer.h",
"intelligibility/intelligibility_utils.cc",
"intelligibility/intelligibility_utils.h",
]
} else {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
}
GN: Prefix WebRTC specific variables with "rtc_" BUG=3441 TESTED=Trybots + Running GN in a Chromium checkout with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using: --args="os=\"android\" cpu_arch=\"arm\"" R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/27379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:36:10 +00:00
if (rtc_prefer_fixed_point) {
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 += [ "WEBRTC_NS_FIXED" ]
} else {
defines += [ "WEBRTC_NS_FLOAT" ]
}
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530) Includes GN changes from https://webrtc-codereview.appspot.com/39249004/ Android changes for JNI were required due to https://codereview.chromium.org/843103003 Other relevant changes: * src/buildtools: 5c5e924..93b3d0a * src/third_party/boringssl/src: d306f16..b180ee9 * src/third_party/icu: 4e3266f..2081ee6 * src/third_party/libvpx: 5cdd302..33bbffe * src/third_party/usrsctp/usrsctplib: 190c8cb..13718c7 * src/tools/gyp: 4d7c139..3464008 * src/tools/swarming_client: bdad118..1b7bfec Details: https://chromium.googlesource.com/chromium/src/+/b0c3ed3..2c3ffb2/DEPS Clang version was not updated in this roll. R=dpranke@chromium.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40079004 Cr-Commit-Position: refs/heads/master@{#8466} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8466 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 19:08:31 +00:00
if (current_cpu == "x86" || current_cpu == "x64") {
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 += [ ":audio_processing_sse2" ]
}
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
if (rtc_build_with_neon) {
GN: 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 += [ ":audio_processing_neon" ]
}
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530) Includes GN changes from https://webrtc-codereview.appspot.com/39249004/ Android changes for JNI were required due to https://codereview.chromium.org/843103003 Other relevant changes: * src/buildtools: 5c5e924..93b3d0a * src/third_party/boringssl/src: d306f16..b180ee9 * src/third_party/icu: 4e3266f..2081ee6 * src/third_party/libvpx: 5cdd302..33bbffe * src/third_party/usrsctp/usrsctplib: 190c8cb..13718c7 * src/tools/gyp: 4d7c139..3464008 * src/tools/swarming_client: bdad118..1b7bfec Details: https://chromium.googlesource.com/chromium/src/+/b0c3ed3..2c3ffb2/DEPS Clang version was not updated in this roll. R=dpranke@chromium.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40079004 Cr-Commit-Position: refs/heads/master@{#8466} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8466 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 19:08:31 +00:00
if (current_cpu == "mipsel") {
What was done was -changed filenames to *.cc -fixed issues with implicit casts causing build errors. All other CL warnings were ignored as the original code is not changed, merely moved. BUG=webrtc:5724 NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1857153002 Cr-Commit-Position: refs/heads/master@{#12307}
2016-04-10 22:38:14 -07:00
sources += [ "aecm/aecm_core_mips.cc" ]
Let Chromium declare the mips_dsp_rev build variable. In https://codereview.chromium.org/883253003, the mips_dsp_rev build variable is added to Chromium's GYP and GN build files. Remove the declarations of mips_dsp_rev from WebRTC's GYP and GN build files. Replace mips_fpu with mips_float_abi and remove the compiler flags that are already set by Chromium. The main review of this was done in https://webrtc-codereview.appspot.com/39779004 but since that CL wasn't created with the right base URL, I made this in order to be able to run WebRTC trybots properly. BUG=446234 TBR=wtc@chromium.org Review URL: https://webrtc-codereview.appspot.com/44549004 Cr-Commit-Position: refs/heads/master@{#8590} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8590 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 09:50:31 +00:00
if (mips_float_abi == "hard") {
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 += [
Moved the AEC C code to be built using C++. In order for the change to be reviewable, the move was made into two steps consisting of the first two patches in this CL. Step 1 (patch set 1): -Changed file types to use .cc -Changed buildfiles to use the new files -Changed C code inclusion to properly match the changed file formats (removed and added extern "C" declarations). -Changed implicit void-> nonvoid casts that are illegal in C++ to be explicit. Step 2 (patch set 2): -Changed all the warnings reported when uploading the CL. -The warnings about formatting of the assembly optimized code were not addressed though. BUG=webrtc:5201 Review URL: https://codereview.webrtc.org/1713923002 Cr-Commit-Position: refs/heads/master@{#11727}
2016-02-23 14:34:59 -08:00
"aec/aec_core_mips.cc",
The FFT functionality in aec_rdft* is based on legacy C code which is not thread-safe in the sense that the rdft_init method can only be run in a single-threaded. Currently, inside WebRTC multiple instances of the audio- processing module are set up which means that the init method may be run concurrently. In order to avoid having to protect the init method with a lock to ensure single-threaded behavior that, this CL places the FFT functionality inside a class so that there is no global component of the FFT functionality. Note that: 1) The nonstandard header for the ooura_fft.cc was copied from the aec_rdft.cc header, and augmented with a description of the changes introduced in this CL. 2) The clang warnings for the ooura_fft_sse2.cc, ooura_fft_neon.cc and ooura_fft_mips.cc were not addressed as this code was kept as it was before this CL 3) Clang-format was run on all files apart from ooura_fft_mips.cc (as that would change the format of the inline assempbly code). Adding bypass of presubmit to avoid code style and header errors caused by the fact that files with legacy code are being renamed. NOPRESUBMIT=true BUG=chromium:638583 Review-Url: https://codereview.webrtc.org/2348213002 Cr-Commit-Position: refs/heads/master@{#14554}
2016-10-06 06:46:20 -07:00
"utility/ooura_fft_mips.cc",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
}
} else {
What was done was -changed filenames to *.cc -fixed issues with implicit casts causing build errors. All other CL warnings were ignored as the original code is not changed, merely moved. BUG=webrtc:5724 NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1857153002 Cr-Commit-Position: refs/heads/master@{#12307}
2016-04-10 22:38:14 -07:00
sources += [ "aecm/aecm_core_c.cc" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
GN: Replace Windows suppressions of warning 4267 with config. This makes the GN configurations easier to read. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2020343003 Cr-Commit-Position: refs/heads/master@{#13006}
2016-06-02 02:09:52 -07:00
# TODO(jschuh): Bug 1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
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 += [
Reland 28629004: adding new AEC dump start interface for chrome. This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here: http://msdn.microsoft.com/en-us/library/ms235460.aspx R=andresp@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, henrikg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7418 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 08:36:56 +00:00
"../../base:rtc_base_approved",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"../../common_audio",
"../../system_wrappers",
]
}
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
rtc_source_set("audio_processing_c") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
sources = [
"agc/legacy/analog_agc.c",
"agc/legacy/analog_agc.h",
"agc/legacy/digital_agc.c",
"agc/legacy/digital_agc.h",
"agc/legacy/gain_control.h",
]
if (rtc_prefer_fixed_point) {
sources += [
"ns/noise_suppression_x.c",
"ns/noise_suppression_x.h",
"ns/nsx_core.c",
"ns/nsx_core.h",
"ns/nsx_defines.h",
]
if (current_cpu == "mipsel") {
sources += [ "ns/nsx_core_mips.c" ]
} else {
sources += [ "ns/nsx_core_c.c" ]
}
} else {
sources += [
"ns/defines.h",
"ns/noise_suppression.c",
"ns/noise_suppression.h",
"ns/ns_core.c",
"ns/ns_core.h",
"ns/windows_private.h",
]
}
deps = [
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
"../..:webrtc_common",
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
"../../base:rtc_base_approved",
"../../common_audio",
"../../system_wrappers",
]
Fixing cross-compiling issues on android arm BUG=webrtc:7042 Review-Url: https://codereview.webrtc.org/2647293006 Cr-Commit-Position: refs/heads/master@{#16265}
2017-01-25 05:34:46 -08:00
if (rtc_build_with_neon) {
deps += [ ":audio_processing_neon_c" ]
}
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
}
GN: Prefix WebRTC specific variables with "rtc_" BUG=3441 TESTED=Trybots + Running GN in a Chromium checkout with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using: --args="os=\"android\" cpu_arch=\"arm\"" R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/27379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:36:10 +00:00
if (rtc_enable_protobuf) {
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
proto_library("audioproc_debug_proto") {
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
sources = [
"debug.proto",
]
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
Update path for audioproc_debug proto output. This will make it align with protoc tools that use the relative path from the project root to the files in the output path. Having this, no hacks will need to be applied downstream. TBR=henrik.lundin@webrtc.org NOTRY=True Review URL: https://codereview.webrtc.org/1673263002 Cr-Commit-Position: refs/heads/master@{#11540}
2016-02-09 08:13:06 -08:00
proto_out_dir = "webrtc/modules/audio_processing"
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
}
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530) Includes GN changes from https://webrtc-codereview.appspot.com/39249004/ Android changes for JNI were required due to https://codereview.chromium.org/843103003 Other relevant changes: * src/buildtools: 5c5e924..93b3d0a * src/third_party/boringssl/src: d306f16..b180ee9 * src/third_party/icu: 4e3266f..2081ee6 * src/third_party/libvpx: 5cdd302..33bbffe * src/third_party/usrsctp/usrsctplib: 190c8cb..13718c7 * src/tools/gyp: 4d7c139..3464008 * src/tools/swarming_client: bdad118..1b7bfec Details: https://chromium.googlesource.com/chromium/src/+/b0c3ed3..2c3ffb2/DEPS Clang version was not updated in this roll. R=dpranke@chromium.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40079004 Cr-Commit-Position: refs/heads/master@{#8466} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8466 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 19:08:31 +00:00
if (current_cpu == "x86" || current_cpu == "x64") {
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("audio_processing_sse2") {
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
# TODO(ehmaldonado): Remove (bugs.webrtc.org/6828)
# Errors on cyclic dependency with :audio_processing if enabled.
check_includes = false
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 = [
Moved the AEC C code to be built using C++. In order for the change to be reviewable, the move was made into two steps consisting of the first two patches in this CL. Step 1 (patch set 1): -Changed file types to use .cc -Changed buildfiles to use the new files -Changed C code inclusion to properly match the changed file formats (removed and added extern "C" declarations). -Changed implicit void-> nonvoid casts that are illegal in C++ to be explicit. Step 2 (patch set 2): -Changed all the warnings reported when uploading the CL. -The warnings about formatting of the assembly optimized code were not addressed though. BUG=webrtc:5201 Review URL: https://codereview.webrtc.org/1713923002 Cr-Commit-Position: refs/heads/master@{#11727}
2016-02-23 14:34:59 -08:00
"aec/aec_core_sse2.cc",
The FFT functionality in aec_rdft* is based on legacy C code which is not thread-safe in the sense that the rdft_init method can only be run in a single-threaded. Currently, inside WebRTC multiple instances of the audio- processing module are set up which means that the init method may be run concurrently. In order to avoid having to protect the init method with a lock to ensure single-threaded behavior that, this CL places the FFT functionality inside a class so that there is no global component of the FFT functionality. Note that: 1) The nonstandard header for the ooura_fft.cc was copied from the aec_rdft.cc header, and augmented with a description of the changes introduced in this CL. 2) The clang warnings for the ooura_fft_sse2.cc, ooura_fft_neon.cc and ooura_fft_mips.cc were not addressed as this code was kept as it was before this CL 3) Clang-format was run on all files apart from ooura_fft_mips.cc (as that would change the format of the inline assempbly code). Adding bypass of presubmit to avoid code style and header errors caused by the fact that files with legacy code are being renamed. NOPRESUBMIT=true BUG=chromium:638583 Review-Url: https://codereview.webrtc.org/2348213002 Cr-Commit-Position: refs/heads/master@{#14554}
2016-10-06 06:46:20 -07:00
"utility/ooura_fft_sse2.cc",
"utility/ooura_fft_tables_neon_sse2.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
]
Fix unknown option '-msse2' warning R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43169004 Cr-Commit-Position: refs/heads/master@{#9016}
2015-04-16 08:57:16 +02:00
if (is_posix) {
cflags = [ "-msse2" ]
}
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
Moved the place for the aec_debug_dump build flag and changed the name to apm_debug_dump Currently, the aec_debug_dump buildflag can and is used to store data in the whole of the audio processing module. Therefore a more appropriate name is apm_debug_dump which also matches the names of the data dumping functionality. This CL makes that name change. The CL also changes the WEBRTC_AEC_DEBUG_DUMP define to WEBRTC_APM_DEBUG_DUMP == 1 Furthermore, this CL moves the buildflag to a more appropriate place. BUG=webrtc:5298 Review-Url: https://codereview.webrtc.org/2300813004 Cr-Commit-Position: refs/heads/master@{#14026}
2016-09-01 08:58:21 -07:00
if (apm_debug_dump) {
defines = [ "WEBRTC_APM_DEBUG_DUMP=1" ]
This CL introduces a new data logging functionality to use for the APM. It allows simple and rapid additions of exploratory data logpoints to use during bug investigations and module performance analysis. The new data logging functionality is also in this CL used to replace the existing data logging functionality present in the AEC. Additional information: As there was an issue with that the build flag for activating this feature was not present in all compilation units that included the feature additional changes were needed. A summary of the changes are -The build files were modified to ensure that the logging build flag always is set to either 0 or 1 for compilation units that include the feature. -Build-time checks in the appropriate places were added to ensure that the above is fulfilled. -The build object was added dynamically to the AEC state as a pointer to ensure that the size of that state is not dependent on whether the logging build flag is set or not. -The constructor of the AEC class needed to be modified in order to construct the logging object. For this a destructor was also needed. -An unused method without any declaration was removed in order to avoid any issues with the logging flag being set to 0 or 1. This CL will be immediately followed with an upcoming CL that replaces the logging in echo_cancellation.cc with the new functionality which will ensure that the logging flag is only used in one place within WebRTC, which in turn will fully ensure that all compilation units that uses the feature also have the flag properly set. BUG=webrtc:5201, webrtc:5298 Review-Url: https://codereview.webrtc.org/1877713002 Cr-Commit-Position: refs/heads/master@{#12607}
2016-05-03 07:01:18 -07:00
} else {
Moved the place for the aec_debug_dump build flag and changed the name to apm_debug_dump Currently, the aec_debug_dump buildflag can and is used to store data in the whole of the audio processing module. Therefore a more appropriate name is apm_debug_dump which also matches the names of the data dumping functionality. This CL makes that name change. The CL also changes the WEBRTC_AEC_DEBUG_DUMP define to WEBRTC_APM_DEBUG_DUMP == 1 Furthermore, this CL moves the buildflag to a more appropriate place. BUG=webrtc:5298 Review-Url: https://codereview.webrtc.org/2300813004 Cr-Commit-Position: refs/heads/master@{#14026}
2016-09-01 08:58:21 -07:00
defines = [ "WEBRTC_APM_DEBUG_DUMP=0" ]
This CL introduces a new data logging functionality to use for the APM. It allows simple and rapid additions of exploratory data logpoints to use during bug investigations and module performance analysis. The new data logging functionality is also in this CL used to replace the existing data logging functionality present in the AEC. Additional information: As there was an issue with that the build flag for activating this feature was not present in all compilation units that included the feature additional changes were needed. A summary of the changes are -The build files were modified to ensure that the logging build flag always is set to either 0 or 1 for compilation units that include the feature. -Build-time checks in the appropriate places were added to ensure that the above is fulfilled. -The build object was added dynamically to the AEC state as a pointer to ensure that the size of that state is not dependent on whether the logging build flag is set or not. -The constructor of the AEC class needed to be modified in order to construct the logging object. For this a destructor was also needed. -An unused method without any declaration was removed in order to avoid any issues with the logging flag being set to 0 or 1. This CL will be immediately followed with an upcoming CL that replaces the logging in echo_cancellation.cc with the new functionality which will ensure that the logging flag is only used in one place within WebRTC, which in turn will fully ensure that all compilation units that uses the feature also have the flag properly set. BUG=webrtc:5201, webrtc:5298 Review-Url: https://codereview.webrtc.org/1877713002 Cr-Commit-Position: refs/heads/master@{#12607}
2016-05-03 07:01:18 -07:00
}
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
}
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
if (rtc_build_with_neon) {
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("audio_processing_neon") {
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
# TODO(ehmaldonado): Remove (bugs.webrtc.org/6828)
# Errors on cyclic dependency with :audio_processing if enabled.
check_includes = false
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 = [
Moved the AEC C code to be built using C++. In order for the change to be reviewable, the move was made into two steps consisting of the first two patches in this CL. Step 1 (patch set 1): -Changed file types to use .cc -Changed buildfiles to use the new files -Changed C code inclusion to properly match the changed file formats (removed and added extern "C" declarations). -Changed implicit void-> nonvoid casts that are illegal in C++ to be explicit. Step 2 (patch set 2): -Changed all the warnings reported when uploading the CL. -The warnings about formatting of the assembly optimized code were not addressed though. BUG=webrtc:5201 Review URL: https://codereview.webrtc.org/1713923002 Cr-Commit-Position: refs/heads/master@{#11727}
2016-02-23 14:34:59 -08:00
"aec/aec_core_neon.cc",
What was done was -changed filenames to *.cc -fixed issues with implicit casts causing build errors. All other CL warnings were ignored as the original code is not changed, merely moved. BUG=webrtc:5724 NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1857153002 Cr-Commit-Position: refs/heads/master@{#12307}
2016-04-10 22:38:14 -07:00
"aecm/aecm_core_neon.cc",
The FFT functionality in aec_rdft* is based on legacy C code which is not thread-safe in the sense that the rdft_init method can only be run in a single-threaded. Currently, inside WebRTC multiple instances of the audio- processing module are set up which means that the init method may be run concurrently. In order to avoid having to protect the init method with a lock to ensure single-threaded behavior that, this CL places the FFT functionality inside a class so that there is no global component of the FFT functionality. Note that: 1) The nonstandard header for the ooura_fft.cc was copied from the aec_rdft.cc header, and augmented with a description of the changes introduced in this CL. 2) The clang warnings for the ooura_fft_sse2.cc, ooura_fft_neon.cc and ooura_fft_mips.cc were not addressed as this code was kept as it was before this CL 3) Clang-format was run on all files apart from ooura_fft_mips.cc (as that would change the format of the inline assempbly code). Adding bypass of presubmit to avoid code style and header errors caused by the fact that files with legacy code are being renamed. NOPRESUBMIT=true BUG=chromium:638583 Review-Url: https://codereview.webrtc.org/2348213002 Cr-Commit-Position: refs/heads/master@{#14554}
2016-10-06 06:46:20 -07:00
"utility/ooura_fft_neon.cc",
"utility/ooura_fft_tables_neon_sse2.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
]
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530) Includes GN changes from https://webrtc-codereview.appspot.com/39249004/ Android changes for JNI were required due to https://codereview.chromium.org/843103003 Other relevant changes: * src/buildtools: 5c5e924..93b3d0a * src/third_party/boringssl/src: d306f16..b180ee9 * src/third_party/icu: 4e3266f..2081ee6 * src/third_party/libvpx: 5cdd302..33bbffe * src/third_party/usrsctp/usrsctplib: 190c8cb..13718c7 * src/tools/gyp: 4d7c139..3464008 * src/tools/swarming_client: bdad118..1b7bfec Details: https://chromium.googlesource.com/chromium/src/+/b0c3ed3..2c3ffb2/DEPS Clang version was not updated in this roll. R=dpranke@chromium.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40079004 Cr-Commit-Position: refs/heads/master@{#8466} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8466 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 19:08:31 +00:00
if (current_cpu != "arm64") {
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
# Enable compilation for the NEON instruction set. This is needed
# since //build/config/arm.gni only enables NEON for iOS, not Android.
# This provides the same functionality as webrtc/build/arm_neon.gypi.
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
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
cflags = [ "-mfpu=neon" ]
Enabling building with NEON on ARM64 This patch enables NEON on ARM64 platform. Passed building both on Android ARMv7 and Android ARM64. BUG=3580 R=andrew@webrtc.org, jridges@masque.com Review URL: https://webrtc-codereview.appspot.com/25069004 Patch from Zhongwei Yao <zhongwei.yao@arm.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7751 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 17:01:40 +00:00
}
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
# Disable LTO on NEON targets due to compiler bug.
# TODO(fdegans): Enable this. See crbug.com/408997.
GN: Prefix WebRTC specific variables with "rtc_" BUG=3441 TESTED=Trybots + Running GN in a Chromium checkout with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using: --args="os=\"android\" cpu_arch=\"arm\"" R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/27379004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:36:10 +00:00
if (rtc_use_lto) {
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
cflags -= [
"-flto",
"-ffat-lto-objects",
]
}
Add a [rtc_]build_with_neon variable to unify conditions. Also consolidate ARM options for gn in an arm_neon_config. R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org Review URL: https://codereview.webrtc.org/1181373004. Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-24 18:25:54 -07:00
deps = [
"../../common_audio",
]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
public_deps = [
":audio_processing_neon_c",
]
This CL introduces a new data logging functionality to use for the APM. It allows simple and rapid additions of exploratory data logpoints to use during bug investigations and module performance analysis. The new data logging functionality is also in this CL used to replace the existing data logging functionality present in the AEC. Additional information: As there was an issue with that the build flag for activating this feature was not present in all compilation units that included the feature additional changes were needed. A summary of the changes are -The build files were modified to ensure that the logging build flag always is set to either 0 or 1 for compilation units that include the feature. -Build-time checks in the appropriate places were added to ensure that the above is fulfilled. -The build object was added dynamically to the AEC state as a pointer to ensure that the size of that state is not dependent on whether the logging build flag is set or not. -The constructor of the AEC class needed to be modified in order to construct the logging object. For this a destructor was also needed. -An unused method without any declaration was removed in order to avoid any issues with the logging flag being set to 0 or 1. This CL will be immediately followed with an upcoming CL that replaces the logging in echo_cancellation.cc with the new functionality which will ensure that the logging flag is only used in one place within WebRTC, which in turn will fully ensure that all compilation units that uses the feature also have the flag properly set. BUG=webrtc:5201, webrtc:5298 Review-Url: https://codereview.webrtc.org/1877713002 Cr-Commit-Position: refs/heads/master@{#12607}
2016-05-03 07:01:18 -07:00
Moved the place for the aec_debug_dump build flag and changed the name to apm_debug_dump Currently, the aec_debug_dump buildflag can and is used to store data in the whole of the audio processing module. Therefore a more appropriate name is apm_debug_dump which also matches the names of the data dumping functionality. This CL makes that name change. The CL also changes the WEBRTC_AEC_DEBUG_DUMP define to WEBRTC_APM_DEBUG_DUMP == 1 Furthermore, this CL moves the buildflag to a more appropriate place. BUG=webrtc:5298 Review-Url: https://codereview.webrtc.org/2300813004 Cr-Commit-Position: refs/heads/master@{#14026}
2016-09-01 08:58:21 -07:00
if (apm_debug_dump) {
defines = [ "WEBRTC_APM_DEBUG_DUMP=1" ]
This CL introduces a new data logging functionality to use for the APM. It allows simple and rapid additions of exploratory data logpoints to use during bug investigations and module performance analysis. The new data logging functionality is also in this CL used to replace the existing data logging functionality present in the AEC. Additional information: As there was an issue with that the build flag for activating this feature was not present in all compilation units that included the feature additional changes were needed. A summary of the changes are -The build files were modified to ensure that the logging build flag always is set to either 0 or 1 for compilation units that include the feature. -Build-time checks in the appropriate places were added to ensure that the above is fulfilled. -The build object was added dynamically to the AEC state as a pointer to ensure that the size of that state is not dependent on whether the logging build flag is set or not. -The constructor of the AEC class needed to be modified in order to construct the logging object. For this a destructor was also needed. -An unused method without any declaration was removed in order to avoid any issues with the logging flag being set to 0 or 1. This CL will be immediately followed with an upcoming CL that replaces the logging in echo_cancellation.cc with the new functionality which will ensure that the logging flag is only used in one place within WebRTC, which in turn will fully ensure that all compilation units that uses the feature also have the flag properly set. BUG=webrtc:5201, webrtc:5298 Review-Url: https://codereview.webrtc.org/1877713002 Cr-Commit-Position: refs/heads/master@{#12607}
2016-05-03 07:01:18 -07:00
} else {
Moved the place for the aec_debug_dump build flag and changed the name to apm_debug_dump Currently, the aec_debug_dump buildflag can and is used to store data in the whole of the audio processing module. Therefore a more appropriate name is apm_debug_dump which also matches the names of the data dumping functionality. This CL makes that name change. The CL also changes the WEBRTC_AEC_DEBUG_DUMP define to WEBRTC_APM_DEBUG_DUMP == 1 Furthermore, this CL moves the buildflag to a more appropriate place. BUG=webrtc:5298 Review-Url: https://codereview.webrtc.org/2300813004 Cr-Commit-Position: refs/heads/master@{#14026}
2016-09-01 08:58:21 -07:00
defines = [ "WEBRTC_APM_DEBUG_DUMP=0" ]
This CL introduces a new data logging functionality to use for the APM. It allows simple and rapid additions of exploratory data logpoints to use during bug investigations and module performance analysis. The new data logging functionality is also in this CL used to replace the existing data logging functionality present in the AEC. Additional information: As there was an issue with that the build flag for activating this feature was not present in all compilation units that included the feature additional changes were needed. A summary of the changes are -The build files were modified to ensure that the logging build flag always is set to either 0 or 1 for compilation units that include the feature. -Build-time checks in the appropriate places were added to ensure that the above is fulfilled. -The build object was added dynamically to the AEC state as a pointer to ensure that the size of that state is not dependent on whether the logging build flag is set or not. -The constructor of the AEC class needed to be modified in order to construct the logging object. For this a destructor was also needed. -An unused method without any declaration was removed in order to avoid any issues with the logging flag being set to 0 or 1. This CL will be immediately followed with an upcoming CL that replaces the logging in echo_cancellation.cc with the new functionality which will ensure that the logging flag is only used in one place within WebRTC, which in turn will fully ensure that all compilation units that uses the feature also have the flag properly set. BUG=webrtc:5201, webrtc:5298 Review-Url: https://codereview.webrtc.org/1877713002 Cr-Commit-Position: refs/heads/master@{#12607}
2016-05-03 07:01:18 -07:00
}
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
rtc_static_library("audio_processing_neon_c") {
Fixing cross-compiling issues on android arm BUG=webrtc:7042 Review-Url: https://codereview.webrtc.org/2647293006 Cr-Commit-Position: refs/heads/master@{#16265}
2017-01-25 05:34:46 -08:00
# TODO(mbonadei): Remove (bugs.webrtc.org/6828)
# Errors on cyclic dependency with :audio_processing_c if enabled.
check_includes = false
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 = [
"ns/nsx_core_neon.c",
]
if (current_cpu != "arm64") {
# Enable compilation for the NEON instruction set. This is needed
# since //build/config/arm.gni only enables NEON for iOS, not Android.
# This provides the same functionality as webrtc/build/arm_neon.gypi.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
}
# Disable LTO on NEON targets due to compiler bug.
# TODO(fdegans): Enable this. See crbug.com/408997.
if (rtc_use_lto) {
cflags -= [
"-flto",
"-ffat-lto-objects",
]
}
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
deps = [
"../../base:rtc_base_approved",
]
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
}
GN: 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
if (rtc_include_tests) {
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
group("audio_processing_tests") {
testonly = true
public_deps = [
":audioproc_test_utils",
":click_annotate",
":nonlinear_beamformer_test",
":transient_suppression_test",
]
if (rtc_enable_intelligibility_enhancer) {
public_deps += [ ":intelligibility_proc" ]
}
if (rtc_enable_protobuf) {
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
public_deps += [
":audioproc_f",
":audioproc_unittest_proto",
":unpack_aecdump",
Reland "C++ porting of the initial python script for conversational speech generation." This CL removes the Python script and adds its C++ porting. The former was in its early stage and it has permanently been removed. This is a reland of https://codereview.webrtc.org/2740063004/ which was reverted. Now the build errors are fixed. BUG=webrtc:7218 Review-Url: https://codereview.webrtc.org/2752793002 Cr-Commit-Position: refs/heads/master@{#17300}
2017-03-18 02:29:13 -07:00
"test/conversational_speech",
Parallel iterator bash script. The Python files are empty ones, just a placeholder for the code they will host. Sample audio files for the APM quality assessment toolbox BUG=webrtc:7218 NOTRY=True Review-Url: https://codereview.webrtc.org/2711923002 Cr-Commit-Position: refs/heads/master@{#16827}
2017-02-24 05:32:21 -08:00
"test/py_quality_assessment",
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
]
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
}
}
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
rtc_source_set("audio_processing_unittests") {
testonly = true
sources = [
"aec/echo_cancellation_unittest.cc",
"aec/system_delay_unittest.cc",
"agc/agc_manager_direct_unittest.cc",
"agc/loudness_histogram_unittest.cc",
"agc/mock_agc.h",
"audio_buffer_unittest.cc",
"beamformer/array_util_unittest.cc",
"beamformer/complex_matrix_unittest.cc",
"beamformer/covariance_matrix_generator_unittest.cc",
"beamformer/matrix_unittest.cc",
"beamformer/mock_nonlinear_beamformer.h",
"config_unittest.cc",
"echo_cancellation_impl_unittest.cc",
"splitting_filter_unittest.cc",
"transient/dyadic_decimator_unittest.cc",
"transient/file_utils.cc",
"transient/file_utils.h",
"transient/file_utils_unittest.cc",
"transient/moving_moments_unittest.cc",
"transient/transient_detector_unittest.cc",
"transient/transient_suppressor_unittest.cc",
"transient/wpd_node_unittest.cc",
"transient/wpd_tree_unittest.cc",
"utility/block_mean_calculator_unittest.cc",
"utility/delay_estimator_unittest.cc",
"vad/gmm_unittest.cc",
"vad/pitch_based_vad_unittest.cc",
"vad/pitch_internal_unittest.cc",
"vad/pole_zero_filter_unittest.cc",
"vad/standalone_vad_unittest.cc",
"vad/vad_audio_proc_unittest.cc",
"vad/vad_circular_buffer_unittest.cc",
"vad/voice_activity_detector_unittest.cc",
]
deps = [
":audio_processing",
":audioproc_test_utils",
"../..:webrtc_common",
"../../base:gtest_prod",
"../../base:rtc_base",
"../../base:rtc_base_approved",
"../../common_audio:common_audio",
"../../system_wrappers:system_wrappers",
"../../test:test_support",
"../audio_coding:neteq_unittest_tools",
"//testing/gmock",
"//testing/gtest",
]
defines = []
if (apm_debug_dump) {
defines += [ "WEBRTC_APM_DEBUG_DUMP=1" ]
} else {
defines += [ "WEBRTC_APM_DEBUG_DUMP=0" ]
}
if (rtc_enable_intelligibility_enhancer) {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
sources += [
"intelligibility/intelligibility_enhancer_unittest.cc",
"intelligibility/intelligibility_utils_unittest.cc",
]
} else {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
}
if (rtc_prefer_fixed_point) {
defines += [ "WEBRTC_AUDIOPROC_FIXED_PROFILE" ]
} else {
defines += [ "WEBRTC_AUDIOPROC_FLOAT_PROFILE" ]
}
if (rtc_enable_protobuf) {
defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
deps += [
GN: Make audio_processing_unittests compile with rtc_enable_protobuf=false BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2647003002 Cr-Commit-Position: refs/heads/master@{#16209}
2017-01-23 04:26:02 -08:00
":audioproc_debug_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
":audioproc_protobuf_utils",
":audioproc_unittest_proto",
]
sources += [
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/adaptive_fir_filter_unittest.cc",
"aec3/aec3_fft_unittest.cc",
"aec3/aec_state_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"aec3/block_framer_unittest.cc",
Adding metrics to AEC3. This CL adds metrics reporting to AEC3. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2722453002 Cr-Commit-Position: refs/heads/master@{#16929}
2017-02-28 22:08:53 -08:00
"aec3/block_processor_metrics_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"aec3/block_processor_unittest.cc",
"aec3/cascaded_biquad_filter_unittest.cc",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/comfort_noise_generator_unittest.cc",
Adding full initial version of delay estimation functionality in echo canceller 3 This CL adds code to the all the delay estimation functionality that is available for the first version of echo canceller 3. The code completes the class EchoPathDelayEstimator. Note that this code does not yet include any handling of clock-drift so there will be upcoming versions of this code. Also note that the CL includes some minor changes in other files for echo canceller 3. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2644123002 Cr-Commit-Position: refs/heads/master@{#16489}
2017-02-08 05:08:56 -08:00
"aec3/decimator_by_4_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"aec3/echo_canceller3_unittest.cc",
Adding second layer of the echo canceller 3 functionality. This CL adds code to the BlockProcessor, which basically constitutes the second layer in echo canceller 3. The CL includes two incomplete classes (EchoRemover and EchoPathDelayEstimator) which will be completed in upcoming CLs. Because of this, some of the unittests are disabled until those are added. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2611223003 Cr-Commit-Position: refs/heads/master@{#16319}
2017-01-27 03:28:19 -08:00
"aec3/echo_path_delay_estimator_unittest.cc",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/echo_path_variability_unittest.cc",
Adding metrics to AEC3. This CL adds metrics reporting to AEC3. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2722453002 Cr-Commit-Position: refs/heads/master@{#16929}
2017-02-28 22:08:53 -08:00
"aec3/echo_remover_metrics_unittest.cc",
Adding second layer of the echo canceller 3 functionality. This CL adds code to the BlockProcessor, which basically constitutes the second layer in echo canceller 3. The CL includes two incomplete classes (EchoRemover and EchoPathDelayEstimator) which will be completed in upcoming CLs. Because of this, some of the unittests are disabled until those are added. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2611223003 Cr-Commit-Position: refs/heads/master@{#16319}
2017-01-27 03:28:19 -08:00
"aec3/echo_remover_unittest.cc",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/erl_estimator_unittest.cc",
"aec3/erle_estimator_unittest.cc",
"aec3/fft_buffer_unittest.cc",
"aec3/fft_data_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"aec3/frame_blocker_unittest.cc",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/main_filter_update_gain_unittest.cc",
Adding full initial version of delay estimation functionality in echo canceller 3 This CL adds code to the all the delay estimation functionality that is available for the first version of echo canceller 3. The code completes the class EchoPathDelayEstimator. Note that this code does not yet include any handling of clock-drift so there will be upcoming versions of this code. Also note that the CL includes some minor changes in other files for echo canceller 3. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2644123002 Cr-Commit-Position: refs/heads/master@{#16489}
2017-02-08 05:08:56 -08:00
"aec3/matched_filter_lag_aggregator_unittest.cc",
"aec3/matched_filter_unittest.cc",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/output_selector_unittest.cc",
"aec3/power_echo_model_unittest.cc",
Adding second layer of the echo canceller 3 functionality. This CL adds code to the BlockProcessor, which basically constitutes the second layer in echo canceller 3. The CL includes two incomplete classes (EchoRemover and EchoPathDelayEstimator) which will be completed in upcoming CLs. Because of this, some of the unittests are disabled until those are added. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2611223003 Cr-Commit-Position: refs/heads/master@{#16319}
2017-01-27 03:28:19 -08:00
"aec3/render_delay_buffer_unittest.cc",
Adding metrics to AEC3. This CL adds metrics reporting to AEC3. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2722453002 Cr-Commit-Position: refs/heads/master@{#16929}
2017-02-28 22:08:53 -08:00
"aec3/render_delay_controller_metrics_unittest.cc",
Adding second layer of the echo canceller 3 functionality. This CL adds code to the BlockProcessor, which basically constitutes the second layer in echo canceller 3. The CL includes two incomplete classes (EchoRemover and EchoPathDelayEstimator) which will be completed in upcoming CLs. Because of this, some of the unittests are disabled until those are added. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2611223003 Cr-Commit-Position: refs/heads/master@{#16319}
2017-01-27 03:28:19 -08:00
"aec3/render_delay_controller_unittest.cc",
Finalization of the first version of EchoCanceller 3 This CL adds the remaining code for the first version of EchoCanceller3. TBR=aleloi@webrtc.org BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2678423005 Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 05:16:26 -08:00
"aec3/render_signal_analyzer_unittest.cc",
"aec3/residual_echo_estimator_unittest.cc",
"aec3/shadow_filter_update_gain_unittest.cc",
"aec3/subtractor_unittest.cc",
"aec3/suppression_filter_unittest.cc",
"aec3/suppression_gain_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"audio_processing_impl_locking_unittest.cc",
"audio_processing_impl_unittest.cc",
"audio_processing_unittest.cc",
GN: Make audio_processing_unittests compile with rtc_enable_protobuf=false BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2647003002 Cr-Commit-Position: refs/heads/master@{#16209}
2017-01-23 04:26:02 -08:00
"beamformer/nonlinear_beamformer_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"echo_cancellation_bit_exact_unittest.cc",
"echo_control_mobile_unittest.cc",
"echo_detector/circular_buffer_unittest.cc",
"echo_detector/mean_variance_estimator_unittest.cc",
"echo_detector/moving_max_unittest.cc",
"echo_detector/normalized_covariance_estimator_unittest.cc",
"gain_control_unittest.cc",
"level_controller/level_controller_unittest.cc",
"level_estimator_unittest.cc",
"low_cut_filter_unittest.cc",
"noise_suppression_unittest.cc",
"residual_echo_detector_unittest.cc",
"rms_level_unittest.cc",
"test/bitexactness_tools.cc",
"test/bitexactness_tools.h",
"test/debug_dump_replayer.cc",
"test/debug_dump_replayer.h",
"test/debug_dump_test.cc",
Adding second layer of the echo canceller 3 functionality. This CL adds code to the BlockProcessor, which basically constitutes the second layer in echo canceller 3. The CL includes two incomplete classes (EchoRemover and EchoPathDelayEstimator) which will be completed in upcoming CLs. Because of this, some of the unittests are disabled until those are added. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2611223003 Cr-Commit-Position: refs/heads/master@{#16319}
2017-01-27 03:28:19 -08:00
"test/echo_canceller_test_tools.cc",
"test/echo_canceller_test_tools.h",
"test/echo_canceller_test_tools_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"test/test_utils.h",
"voice_detection_unittest.cc",
]
}
if ((!build_with_chromium || is_win) && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
rtc_source_set("audio_processing_perf_tests") {
# Has problems with autogenerated targets on Android and iOS
# Dependency chain (there may also be others):
# //webrtc/modules/audio_processing:audio_processing_perf_tests -->
# //webrtc/modules:modules_unittests --[private]-->
# //webrtc/modules:modules_unittests_apk -->
# //webrtc/modules:modules_unittests_apk__create -->
# //webrtc/modules:modules_unittests_apk__create__finalize -->
# //webrtc/modules:modules_unittests_apk__create__package --[private]-->
# //webrtc/modules:_modules_unittests__library
check_includes = false
testonly = true
sources = [
"audio_processing_performance_unittest.cc",
"level_controller/level_controller_complexity_unittest.cc",
"residual_echo_detector_complexity_unittest.cc",
]
deps = [
":audio_processing",
":audioproc_test_utils",
"//testing/gtest",
]
if (rtc_enable_intelligibility_enhancer) {
defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
} else {
defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
}
}
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
if (rtc_enable_protobuf) {
rtc_executable("unpack_aecdump") {
testonly = true
sources = [
"test/unpack.cc",
]
Adding GN targets for audioproc, audioproc_f and unpack_aecdump BUG=webrtc:6171 NOTRY=True Review-Url: https://codereview.webrtc.org/2222643002 Cr-Commit-Position: refs/heads/master@{#13667}
2016-08-07 23:19:30 -07:00
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
deps = [
":audio_processing",
":audioproc_debug_proto",
":audioproc_protobuf_utils",
":audioproc_test_utils",
"../..:webrtc_common",
"../../base:rtc_base_approved",
"../../common_audio",
"../../system_wrappers:system_wrappers_default",
"//third_party/gflags:gflags",
]
} # unpack_aecdump
rtc_executable("audioproc_f") {
testonly = true
sources = [
"test/aec_dump_based_simulator.cc",
"test/aec_dump_based_simulator.h",
"test/audio_processing_simulator.cc",
"test/audio_processing_simulator.h",
"test/audioproc_float.cc",
"test/wav_based_simulator.cc",
"test/wav_based_simulator.h",
]
Adding GN targets for audioproc, audioproc_f and unpack_aecdump BUG=webrtc:6171 NOTRY=True Review-Url: https://codereview.webrtc.org/2222643002 Cr-Commit-Position: refs/heads/master@{#13667}
2016-08-07 23:19:30 -07:00
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false. This is a first step towards it. I plan to refactor modules_unittests before continuing with this. BUG=webrtc:6626 NOTRY=True Review-Url: https://codereview.webrtc.org/2626163004 Cr-Commit-Position: refs/heads/master@{#16109}
2017-01-17 02:37:34 -08:00
deps = [
":audio_processing",
":audioproc_debug_proto",
":audioproc_protobuf_utils",
":audioproc_test_utils",
"../../base:rtc_base_approved",
"../../common_audio:common_audio",
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
"//testing/gtest",
"//third_party/gflags:gflags",
]
} # audioproc_f
}
Adding GN targets for audioproc, audioproc_f and unpack_aecdump BUG=webrtc:6171 NOTRY=True Review-Url: https://codereview.webrtc.org/2222643002 Cr-Commit-Position: refs/heads/master@{#13667}
2016-08-07 23:19:30 -07:00
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_source_set("audioproc_test_utils") {
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 = [
Add webrtc_perf_tests to BUILD.gn Updated the sources in audio_processing:audioproc_test_utils to match the configuration on "webrtc/modules/audio_processing/audio_processing_tests.gypi" Removed audio_buffer_tools from modules_unittests to match the gyp file. BUG=webrtc:6041 Review-Url: https://codereview.webrtc.org/2178963002 Cr-Commit-Position: refs/heads/master@{#13541}
2016-07-27 08:14:32 -07:00
"test/audio_buffer_tools.cc",
"test/audio_buffer_tools.h",
Added a perf test for the residual echo detector. This perf tests the echo detector in 3 scenarios: standalone, as part of APM with only the echo detector enabled and as part of a normally configured APM. BUG=webrtc:6525 Review-Url: https://codereview.webrtc.org/2517523003 Cr-Commit-Position: refs/heads/master@{#15224}
2016-11-24 04:17:28 -08:00
"test/performance_timer.cc",
"test/performance_timer.h",
"test/simulator_buffers.cc",
"test/simulator_buffers.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
"test/test_utils.cc",
"test/test_utils.h",
]
deps = [
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
":audio_processing",
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
"../../base:rtc_base_approved",
"../../common_audio",
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
"../../system_wrappers:system_wrappers",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("transient_suppression_test") {
GN build rules for four audio processing test executables click_annotate, intelligibility_proc, nonlinear_beamformer_test, and transient_suppression_test. This is a re-land of https://codereview.webrtc.org/2267403003 BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2273783004 Cr-Commit-Position: refs/heads/master@{#13913}
2016-08-25 02:32:43 -07:00
testonly = true
sources = [
"transient/file_utils.cc",
"transient/file_utils.h",
"transient/transient_suppression_test.cc",
]
deps = [
":audio_processing",
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
"../..:webrtc_common",
"../../common_audio:common_audio",
GN build rules for four audio processing test executables click_annotate, intelligibility_proc, nonlinear_beamformer_test, and transient_suppression_test. This is a re-land of https://codereview.webrtc.org/2267403003 BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2273783004 Cr-Commit-Position: refs/heads/master@{#13913}
2016-08-25 02:32:43 -07:00
"../../system_wrappers:metrics_default",
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
"../../system_wrappers:system_wrappers",
GN build rules for four audio processing test executables click_annotate, intelligibility_proc, nonlinear_beamformer_test, and transient_suppression_test. This is a re-land of https://codereview.webrtc.org/2267403003 BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2273783004 Cr-Commit-Position: refs/heads/master@{#13913}
2016-08-25 02:32:43 -07:00
"../../test:test_support",
"//testing/gtest",
"//third_party/gflags",
]
}
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("click_annotate") {
GN build rules for four audio processing test executables click_annotate, intelligibility_proc, nonlinear_beamformer_test, and transient_suppression_test. This is a re-land of https://codereview.webrtc.org/2267403003 BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2273783004 Cr-Commit-Position: refs/heads/master@{#13913}
2016-08-25 02:32:43 -07:00
testonly = true
sources = [
"transient/click_annotate.cc",
"transient/file_utils.cc",
"transient/file_utils.h",
]
deps = [
":audio_processing",
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
"../..:webrtc_common",
GN build rules for four audio processing test executables click_annotate, intelligibility_proc, nonlinear_beamformer_test, and transient_suppression_test. This is a re-land of https://codereview.webrtc.org/2267403003 BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2273783004 Cr-Commit-Position: refs/heads/master@{#13913}
2016-08-25 02:32:43 -07:00
"../../system_wrappers:metrics_default",
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
"../../system_wrappers:system_wrappers",
GN build rules for four audio processing test executables click_annotate, intelligibility_proc, nonlinear_beamformer_test, and transient_suppression_test. This is a re-land of https://codereview.webrtc.org/2267403003 BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2273783004 Cr-Commit-Position: refs/heads/master@{#13913}
2016-08-25 02:32:43 -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("nonlinear_beamformer_test") {
GN build rules for four audio processing test executables click_annotate, intelligibility_proc, nonlinear_beamformer_test, and transient_suppression_test. This is a re-land of https://codereview.webrtc.org/2267403003 BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2273783004 Cr-Commit-Position: refs/heads/master@{#13913}
2016-08-25 02:32:43 -07:00
testonly = true
sources = [
"beamformer/nonlinear_beamformer_test.cc",
]
deps = [
":audio_processing",
":audioproc_test_utils",
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
"../../base:rtc_base_approved",
"../../common_audio:common_audio",
GN build rules for four audio processing test executables click_annotate, intelligibility_proc, nonlinear_beamformer_test, and transient_suppression_test. This is a re-land of https://codereview.webrtc.org/2267403003 BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2273783004 Cr-Commit-Position: refs/heads/master@{#13913}
2016-08-25 02:32:43 -07:00
"../../system_wrappers:metrics_default",
"//third_party/gflags",
]
}
Deactivated the intelligibility enhancement functionality by default NOTRY=true BUG= Review-Url: https://codereview.webrtc.org/2272423003 Cr-Commit-Position: refs/heads/master@{#13937}
2016-08-26 07:16:04 -07:00
if (rtc_enable_intelligibility_enhancer) {
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("intelligibility_proc") {
Deactivated the intelligibility enhancement functionality by default NOTRY=true BUG= Review-Url: https://codereview.webrtc.org/2272423003 Cr-Commit-Position: refs/heads/master@{#13937}
2016-08-26 07:16:04 -07:00
testonly = true
sources = [
"intelligibility/test/intelligibility_proc.cc",
]
deps = [
":audio_processing",
":audioproc_test_utils",
"../../system_wrappers:metrics_default",
"../../test:test_support",
"//testing/gtest",
"//third_party/gflags",
]
GN build rules for four audio processing test executables click_annotate, intelligibility_proc, nonlinear_beamformer_test, and transient_suppression_test. This is a re-land of https://codereview.webrtc.org/2267403003 BUG=webrtc:5949 Review-Url: https://codereview.webrtc.org/2273783004 Cr-Commit-Position: refs/heads/master@{#13913}
2016-08-25 02:32:43 -07:00
}
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (rtc_enable_protobuf) {
proto_library("audioproc_unittest_proto") {
sources = [
"test/unittest.proto",
]
Restructure protobuf targets to fit with downstream requirements. The paths of the protobuf output files needs to match the actual tree path in order for Bazel builds to work. BUG=webrtc:6412 NOTRY=True Review-Url: https://codereview.webrtc.org/2716413004 Cr-Commit-Position: refs/heads/master@{#16912}
2017-02-28 08:01:46 -08:00
proto_out_dir = "webrtc/modules/audio_processing/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
}
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("audioproc_protobuf_utils") {
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
sources = [
"test/protobuf_utils.cc",
"test/protobuf_utils.h",
]
deps = [
GN: Add "//build/config/sanitizers:deps" as a dependency to executable targets. When the sanitizer bots are switched to GN, this needs to be included as a dependency so that the executables can be compiled. BUG=webrtc:6215 NOTRY=True Review-Url: https://codereview.webrtc.org/2250893003 Cr-Commit-Position: refs/heads/master@{#13829}
2016-08-19 02:11:07 -07:00
":audioproc_debug_proto",
Refactor webrtc/modules/audio_processing for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/audio_processing/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2558813003 Cr-Commit-Position: refs/heads/master@{#15475}
2016-12-08 04:10:01 -08:00
"../..:webrtc_common",
"../../base:rtc_base_approved",
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
]
}
}
}
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