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GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../../webrtc.gni")
Update gn files to support Mozilla build Bug: webrtc:8670 No-Presubmit: true Change-Id: I085dc63daa8274b5068540cbf56b6330f40643fa Reviewed-on: https://webrtc-review.googlesource.com/38920 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21624}
2018-01-15 10:20:00 -05:00
if (!build_with_mozilla) {
import("//third_party/protobuf/proto_library.gni")
}
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
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
}
Clean up duplication in APM gn file. I realized I could use configs to fix some duplication that I partially introduced. Verified APM_DEBUG_DUMP is set appropriately by looking at the compiler command line. Bug: webrtc:6828 Change-Id: Ia990e2721546d65639567cd3ab788439e328c5da Reviewed-on: https://webrtc-review.googlesource.com/34642 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21349}
2017-12-19 09:45:24 +01:00
config("apm_debug_dump") {
if (apm_debug_dump) {
defines = [ "WEBRTC_APM_DEBUG_DUMP=1" ]
} else {
defines = [ "WEBRTC_APM_DEBUG_DUMP=0" ]
}
}
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") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
Clean up duplication in APM gn file. I realized I could use configs to fix some duplication that I partially introduced. Verified APM_DEBUG_DUMP is set appropriately by looking at the compiler command line. Bug: webrtc:6828 Change-Id: Ia990e2721546d65639567cd3ab788439e328c5da Reviewed-on: https://webrtc-review.googlesource.com/34642 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21349}
2017-12-19 09:45:24 +01:00
configs += [ ":apm_debug_dump" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
sources = [
"audio_buffer.cc",
"audio_buffer.h",
"audio_processing_impl.cc",
"audio_processing_impl.h",
"common.h",
"echo_cancellation_impl.cc",
"echo_cancellation_impl.h",
Add AEC proxies for simple deprecation of AEC configurability. Some changes need access to both the APM interface and the AECs, hence we can't make the changes inside the AECs themselves. The proxies also make it easy to drop support for individual parts of the interfaces one at a time. Bug: webrtc:9535 Change-Id: I3398e1182157f7d8b1e4c455060b830b61c20dd9 Reviewed-on: https://webrtc-review.googlesource.com/94500 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24317}
2018-08-16 10:54:07 +02:00
"echo_cancellation_proxy.cc",
"echo_cancellation_proxy.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"echo_control_mobile_impl.cc",
"echo_control_mobile_impl.h",
Add AEC proxies for simple deprecation of AEC configurability. Some changes need access to both the APM interface and the AECs, hence we can't make the changes inside the AECs themselves. The proxies also make it easy to drop support for individual parts of the interfaces one at a time. Bug: webrtc:9535 Change-Id: I3398e1182157f7d8b1e4c455060b830b61c20dd9 Reviewed-on: https://webrtc-review.googlesource.com/94500 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24317}
2018-08-16 10:54:07 +02:00
"echo_control_mobile_proxy.cc",
"echo_control_mobile_proxy.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",
Add FixedGainController and move GainController2 in APM. The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 11:54:07 +01:00
"gain_controller2.cc",
"gain_controller2.h",
Merge :audio_processing and :aec_dump_interface. Merges the two targets in modules/audio_processing and removes some redundant code. This enables not writing a bunch of redundant code in https://webrtc-review.googlesource.com/c/src/+/70502 ':audio_processing' did depend on ':aec_dump_interface'. 'modules/audio_processing/aec_dump' did depend on 'aec_dump_interface' but not ':audio_processing'. Having the AecDump implementation not depending on 'audio_processing' allows to have faster compilation time and reduces the dependencies. However, maintaining such a decoupling makes APM and AecDump client code more complex. NOTRY=true # want this in and 'ios_api_framework' seems stuck. Bug: webrtc:7404 Change-Id: I75a5f234591014ac42d52bc1a36526072f5be89c Reviewed-on: https://webrtc-review.googlesource.com/76603 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23244}
2018-05-15 13:13:22 +02:00
"include/aec_dump.cc",
"include/aec_dump.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",
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 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",
"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 = [
Reland: Fix all circular deps in audio_processing (but one). Arguably we should add a few more targets, for instance a utility target, but I tried to create as few targets as possible here based on the current usage. Bug: webrtc:6828 Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8 Reviewed-on: https://webrtc-review.googlesource.com/28301 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21199}
2017-12-11 10:28:16 +01:00
":apm_logging",
Add FixedGainController and move GainController2 in APM. The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 11:54:07 +01:00
":audio_frame_view",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
":audio_generator_interface",
Stop using public_deps in modules/audio_processing. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: Ib44266389e6f08a77bd92cffd1eba166147687f4 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29822 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21106}
2017-12-05 15:07:16 +01:00
":audio_processing_c",
Use the new AudioProcessing statistics everywhere. The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 17:29:59 +01:00
":audio_processing_statistics",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
":gain_control_interface",
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
"../..:webrtc_common",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-16 13:43:49 +01:00
"../../api/audio:aec3_config",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-16 13:43:49 +01:00
"../../api/audio:echo_control",
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",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Move fft4g to proper third_party directory Bug: webrtc:8366 Change-Id: I98d3ae56a1d14b3ecacd85a4b3d234e215c8bc58 Reviewed-on: https://webrtc-review.googlesource.com/85642 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24103}
2018-07-25 16:56:18 +02:00
"../../common_audio/third_party/fft4g:fft4g",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../../rtc_base:deprecation",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:gtest_prod",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../../rtc_base:safe_minmax",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:sanitizer",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:cpu_features_api",
Enforcing a stream delay of 0 to be assumed in the AEC on Chrome OS This CL forces the AEC2 to assume a stream delay of 0, thereby avoiding that the incorrect stream delays reported on Chrome OS causes echo issues. Bug: chromium:797274, chromium:797272 Change-Id: I10f295c9f1d735622c55fc56be99a14c6cdd88a2 Reviewed-on: https://webrtc-review.googlesource.com/36081 Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21432}
2017-12-22 16:13:57 +01:00
"../../system_wrappers:field_trial_api",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:metrics_api",
Split aec and aecm into separate build targets This clarifies dependencies and makes it easier to customize builds for different binaries. Also adds BUILD files in aec/ and aecm/. Moves unit tests to their own target, which subjects them to Chromium Clang style checks. The CL contains a fix for a thusly induced warning. Bug: webrtc:9488 Change-Id: I77b680b42a4dccc5f025005e0890f60b4eaf2961 Reviewed-on: https://webrtc-review.googlesource.com/87304 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23887}
2018-07-09 14:28:15 +02:00
"aec:aec",
"aec:aec_core",
"aecm:aecm_core",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"agc",
"agc:agc_legacy_c",
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
"agc2:adaptive_digital",
"agc2:fixed_digital",
Activate the pre-amplifier in AudioProcessing. It's a module for applying a gain to the capture signal. The gain is the first processing step in APM. After this CL, these two features work: * The PreAmplifier can be activated with AudioProcessing::Config::pre_amplifier * The PreApmlifier can be controlled after APM creation by AudioProcessing::SetRuntimeSetting. What's left is a change to AecDumps and to AecDump-replay. NOTRY=True # 1-line change, tests just passed. Bug: webrtc:9138 Change-Id: I85b3af511695b0a9cec2eed6fee7f05080305e1d Reviewed-on: https://webrtc-review.googlesource.com/69811 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22881}
2018-04-16 16:31:22 +02:00
"agc2:gain_applier",
Made modules/audio_processing/vad its own target. WHAT: made a BUILD.gn with library and tests in the Audio Processing Module Voice Activity Detector directory. Updated depending code. Fixed a Clang warning. WHY: to make it possible for a target to depend on just the VAD and not the whole APM. There are other benefits: * Sometimes faster compilation. * The VAD takes up 28000 bytes of libjingle_peerconnection_so.so. Making a peerconnection shared object file without the VAD has to be done in steps. The first step is a custom target for the VAD. Hence this Cl. Change-Id: Iea0207a0b5979db26baaf46b24beaefbb1c431af BUG: webrtc:5716, webrtc:7494 Reviewed-on: https://webrtc-review.googlesource.com/47521 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21893}
2018-02-05 11:15:23 +01:00
"vad",
Replace rtc::Optional with absl::optional in modules/audio processing This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_processing' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0 Reviewed-on: https://webrtc-review.googlesource.com/83982 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23653}
2018-06-19 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
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_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" ]
}
deps += [
"../../common_audio",
Fix circular deps in common_audio. This makes it easier to import cleanly downstream, and makes it a lot easier to reason about. Bug: webrtc:6828 Change-Id: I953e129de73053f8619333fe7e318b36e3a1fffa Reviewed-on: https://webrtc-review.googlesource.com/22722 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20727}
2017-11-17 11:04:15 +01:00
"../../common_audio:fir_filter",
"../../common_audio:fir_filter_factory",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"../../system_wrappers",
]
}
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
rtc_source_set("gain_control_interface") {
sources = [
"include/gain_control.h",
]
}
Use the new AudioProcessing statistics everywhere. The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 17:29:59 +01:00
rtc_source_set("audio_processing_statistics") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
Use the new AudioProcessing statistics everywhere. The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 17:29:59 +01:00
sources = [
"include/audio_processing_statistics.cc",
"include/audio_processing_statistics.h",
]
deps = [
Replace rtc::Optional with absl::optional in modules/audio processing This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_processing' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0 Reviewed-on: https://webrtc-review.googlesource.com/83982 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23653}
2018-06-19 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Use the new AudioProcessing statistics everywhere. The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 17:29:59 +01:00
]
}
Add FixedGainController and move GainController2 in APM. The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 11:54:07 +01:00
rtc_source_set("audio_frame_view") {
sources = [
"include/audio_frame_view.h",
]
deps = [
"../../api:array_view",
]
}
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
rtc_source_set("audio_generator_interface") {
visibility = [ "*" ]
sources = [
"include/audio_generator.h",
]
deps = [
":audio_frame_view",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
]
}
rtc_source_set("audio_generator_factory") {
visibility = [ "*" ]
sources = [
"include/audio_generator_factory.cc",
"include/audio_generator_factory.h",
]
deps = [
":audio_generator_interface",
":file_audio_generator",
"../../common_audio:common_audio",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
]
}
rtc_source_set("file_audio_generator") {
sources = [
"audio_generator/file_audio_generator.cc",
"audio_generator/file_audio_generator.h",
]
deps = [
":audio_generator_interface",
"../../common_audio:common_audio",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers: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.
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
sources = []
Split targets mixing .c and .cc sources. The Bazel build format doesn't support having separate lists of compilation flags for C and C++; it just has a single copts list for cc_library: https://bazel.build/versions/master/docs/be/c-cpp.html#cc_binary.copts This makes it hard to convert our GN targets to Bazel when there are compiler warnings that aren't supported for C (like -Woverloaded-virtual being added in bugs.webrtc.org/6653). The solution for this is to move all .c files to their own targets and remove C++-only compiler flags during conversion. New targets: //webrtc/common_audio:common_audio_c //webrtc/common_audio:common_audio_neon_c //webrtc/modules/audio_coding:g711_c //webrtc/modules/audio_coding:g722_c //webrtc/modules/audio_coding:ilbc_c //webrtc/modules/audio_coding:isac_c //webrtc/modules/audio_coding:isac_fix_c //webrtc/modules/audio_coding:isac_test_util //webrtc/modules/audio_coding:pcm16b_c //webrtc/modules/audio_coding:webrtc_opusj_c //webrtc/modules/audio_device:mac_portaudio //webrtc/modules/audio_procssing:audio_processing_c //webrtc/modules/audio_procssing:audio_processing_neon_c This CL also adds a PRESUBMIT.py check that will throw an error if targets are mixing .c and .cc files, to preven this from regressing. BUG=webrtc:6653 NOTRY=True Review-Url: https://codereview.webrtc.org/2550563003 Cr-Commit-Position: refs/heads/master@{#15433}
2016-12-05 22:47:46 -08:00
if (rtc_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
"../../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Move fft4g to proper third_party directory Bug: webrtc:8366 Change-Id: I98d3ae56a1d14b3ecacd85a4b3d234e215c8bc58 Reviewed-on: https://webrtc-review.googlesource.com/85642 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24103}
2018-07-25 16:56:18 +02:00
"../../common_audio/third_party/fft4g:fft4g",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers:cpu_features_api",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"agc:agc_legacy_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
]
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) {
Inlined audio_processing_neon_c. This solves a circular dep and eliminates a target. This means we will apply neon copts to some files that weren't before, but I don't think that is a problem. Bug: webrtc:6828,webrtc:7042 Change-Id: I3bb656ba5b13d6104b519c2dbf6a4b2814575b87 Reviewed-on: https://webrtc-review.googlesource.com/34183 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21330}
2017-12-18 11:21:14 +01:00
sources += [ "ns/nsx_core_neon.c" ]
if (current_cpu != "arm64") {
Explicitly add -mfpu=neon to all targets that use NEON Remove obsolete comment about Chromium not defining NEON for Android. Semi-related fix: don't use `rtc_remove_configs` directly, `suppressed_configs` is the "public interface". Bug: webrtc:9579 Change-Id: I512628feb462a29432f1356cfef00efe1ddaf84f Reviewed-on: https://webrtc-review.googlesource.com/91761 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24165}
2018-08-01 12:18:05 +02:00
# Enable compilation for the NEON instruction set.
Inlined audio_processing_neon_c. This solves a circular dep and eliminates a target. This means we will apply neon copts to some files that weren't before, but I don't think that is a problem. Bug: webrtc:6828,webrtc:7042 Change-Id: I3bb656ba5b13d6104b519c2dbf6a4b2814575b87 Reviewed-on: https://webrtc-review.googlesource.com/34183 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21330}
2017-12-18 11:21:14 +01:00
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",
]
}
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
}
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
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
proto_out_dir = "modules/audio_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
}
}
Reland: Fix all circular deps in audio_processing (but one). Arguably we should add a few more targets, for instance a utility target, but I tried to create as few targets as possible here based on the current usage. Bug: webrtc:6828 Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8 Reviewed-on: https://webrtc-review.googlesource.com/28301 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21199}
2017-12-11 10:28:16 +01:00
rtc_source_set("apm_logging") {
Clean up duplication in APM gn file. I realized I could use configs to fix some duplication that I partially introduced. Verified APM_DEBUG_DUMP is set appropriately by looking at the compiler command line. Bug: webrtc:6828 Change-Id: Ia990e2721546d65639567cd3ab788439e328c5da Reviewed-on: https://webrtc-review.googlesource.com/34642 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21349}
2017-12-19 09:45:24 +01:00
configs += [ ":apm_debug_dump" ]
Reland: Fix all circular deps in audio_processing (but one). Arguably we should add a few more targets, for instance a utility target, but I tried to create as few targets as possible here based on the current usage. Bug: webrtc:6828 Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8 Reviewed-on: https://webrtc-review.googlesource.com/28301 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21199}
2017-12-11 10:28:16 +01:00
sources = [
"logging/apm_data_dumper.cc",
"logging/apm_data_dumper.h",
]
deps = [
"../../api:array_view",
"../../common_audio:common_audio",
"../../rtc_base:rtc_base_approved",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../../rtc_base:stringutils",
Reland: Fix all circular deps in audio_processing (but one). Arguably we should add a few more targets, for instance a utility target, but I tried to create as few targets as possible here based on the current usage. Bug: webrtc:6828 Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8 Reviewed-on: https://webrtc-review.googlesource.com/28301 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21199}
2017-12-11 10:28:16 +01:00
]
defines = []
}
Revert "Fix all circular deps in audio_processing (but one)." This reverts commit 0af8370cb38b0b0f35f4ed4ec4237d0e6c7d59da. Reason for revert: Breaks downstream Original change's description: > Fix all circular deps in audio_processing (but one). > > Arguably we should add a few more targets, for instance a utility > target, but I tried to create as few targets as possible here based on > the current usage. > > Bug: webrtc:6828 > Change-Id: If2740de2e4374eeae64b3d7599a52bb051594c6a > Reviewed-on: https://webrtc-review.googlesource.com/28020 > Reviewed-by: Per Åhgren <peah@webrtc.org> > Commit-Queue: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21025} TBR=phoglund@webrtc.org,peah@webrtc.org Change-Id: I423f027f6919cf4eb44b4e08c7cb38f0506ad0d7 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:6828 Reviewed-on: https://webrtc-review.googlesource.com/28940 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21027}
2017-12-04 10:11:11 +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) {
Add missing files to audio_processing. Bug: webrtc:7621 Change-Id: I2cab764232fc4e084ed8a489f4cf3a3ac562c894 Reviewed-on: https://webrtc-review.googlesource.com/34658 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21461}
2017-12-19 16:44:45 +01:00
rtc_source_set("mocks") {
testonly = true
sources = [
"include/mock_audio_processing.h",
]
deps = [
":audio_processing",
":audio_processing_statistics",
"../../test:test_support",
]
}
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
group("audio_processing_tests") {
testonly = true
Stop using public_deps in modules/audio_processing. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: Ib44266389e6f08a77bd92cffd1eba166147687f4 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29822 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21106}
2017-12-05 15:07:16 +01:00
deps = [
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
":audioproc_test_utils",
":click_annotate",
":transient_suppression_test",
]
if (rtc_enable_protobuf) {
Stop using public_deps in modules/audio_processing. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: Ib44266389e6f08a77bd92cffd1eba166147687f4 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29822 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21106}
2017-12-05 15:07:16 +01:00
deps += [
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
":audioproc_f",
":audioproc_unittest_proto",
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
"aec_dump:aec_dump_unittests",
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
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
Clean up duplication in APM gn file. I realized I could use configs to fix some duplication that I partially introduced. Verified APM_DEBUG_DUMP is set appropriately by looking at the compiler command line. Bug: webrtc:6828 Change-Id: Ia990e2721546d65639567cd3ab788439e328c5da Reviewed-on: https://webrtc-review.googlesource.com/34642 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21349}
2017-12-19 09:45:24 +01:00
configs += [ ":apm_debug_dump" ]
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
sources = [
"audio_buffer_unittest.cc",
AGC2-fixed-digital: Level Estimator This CL adds the Level Estimator of the new gain controller. The Level Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub frames. We take the maximal sample values in every sub frame. We then apply attack/decay smoothing. This is the final level estimate. The results will be used with InterpolatedGainCurve (see this CL https://webrtc-review.googlesource.com/c/src/+/51920). For every level estimate value, we look up a gain with InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to the signal. Bug: webrtc:7949 Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1 Reviewed-on: https://webrtc-review.googlesource.com/52381 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22054}
2018-02-16 12:39:00 +01:00
"audio_frame_view_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
"config_unittest.cc",
"echo_cancellation_impl_unittest.cc",
AGC2-fixed-digital: Level Estimator This CL adds the Level Estimator of the new gain controller. The Level Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub frames. We take the maximal sample values in every sub frame. We then apply attack/decay smoothing. This is the final level estimate. The results will be used with InterpolatedGainCurve (see this CL https://webrtc-review.googlesource.com/c/src/+/51920). For every level estimate value, we look up a gain with InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to the signal. Bug: webrtc:7949 Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1 Reviewed-on: https://webrtc-review.googlesource.com/52381 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22054}
2018-02-16 12:39:00 +01:00
"gain_controller2_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
"splitting_filter_unittest.cc",
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
"test/fake_recording_device_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
"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",
]
deps = [
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
":analog_mic_simulation",
Reland: Fix all circular deps in audio_processing (but one). Arguably we should add a few more targets, for instance a utility target, but I tried to create as few targets as possible here based on the current usage. Bug: webrtc:6828 Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8 Reviewed-on: https://webrtc-review.googlesource.com/28301 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21199}
2017-12-11 10:28:16 +01:00
":apm_logging",
AGC2-fixed-digital: Level Estimator This CL adds the Level Estimator of the new gain controller. The Level Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub frames. We take the maximal sample values in every sub frame. We then apply attack/decay smoothing. This is the final level estimate. The results will be used with InterpolatedGainCurve (see this CL https://webrtc-review.googlesource.com/c/src/+/51920). For every level estimate value, we look up a gain with InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to the signal. Bug: webrtc:7949 Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1 Reviewed-on: https://webrtc-review.googlesource.com/52381 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22054}
2018-02-16 12:39:00 +01:00
":audio_frame_view",
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",
":audioproc_test_utils",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
":file_audio_generator_unittests",
Add missing files to audio_processing. Bug: webrtc:7621 Change-Id: I2cab764232fc4e084ed8a489f4cf3a3ac562c894 Reviewed-on: https://webrtc-review.googlesource.com/34658 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21461}
2017-12-19 16:44:45 +01:00
":mocks",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../..:webrtc_common",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-16 13:43:49 +01:00
"../../api/audio:aec3_config",
Move EchoCanceller3Factory to api/auido The AEC3 factory is now part of the WebRTC API. Bug: webrtc:8844 Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e Reviewed-on: https://webrtc-review.googlesource.com/57141 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22204}
2018-02-27 13:58:45 +01:00
"../../api/audio:aec3_factory",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../common_audio:common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../../common_audio:common_audio_c",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:gtest_prod",
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../../rtc_base:safe_minmax",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
Move file_wrapper.h to rtc_base/system/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 Change-Id: I440974da4d347b09ff042478720d7983056b62b9 Reviewed-on: https://webrtc-review.googlesource.com/21226 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22579}
2018-03-23 10:39:34 +01:00
"../../rtc_base/system:file_wrapper",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
"../../system_wrappers:cpu_features_api",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:test_support",
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
"../audio_coding:neteq_input_audio_tools",
Split aec and aecm into separate build targets This clarifies dependencies and makes it easier to customize builds for different binaries. Also adds BUILD files in aec/ and aecm/. Moves unit tests to their own target, which subjects them to Chromium Clang style checks. The CL contains a fix for a thusly induced warning. Bug: webrtc:9488 Change-Id: I77b680b42a4dccc5f025005e0890f60b4eaf2961 Reviewed-on: https://webrtc-review.googlesource.com/87304 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23887}
2018-07-09 14:28:15 +02:00
"aec:aec_core",
"aec:aec_unittests",
MockAecDump and integration tests between AecDump and AudioProcessing This CL adds a MockAecDump and integration tests that inject the mock into AudioProcessingImpl. The tests check the call pattern between AudioProcessingImpl and AecDump. The existing tests ApmTest.* and DebugDumpTest.* (not touched by this CL) check that the data written by AecDumpImpl is valid. The tests check that the protobuf-writing methods for the different protobuf message types in audio_processing/debug.proto are indeed called for the different modes of AudioProcessingImpl operation. BUG=webrtc:7404 Review-Url: https://codereview.webrtc.org/2888533005 Cr-Commit-Position: refs/heads/master@{#18501}
2017-06-08 08:12:46 -07:00
"aec_dump:mock_aec_dump_unittests",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"agc:agc_unittests",
Level estimation and saturation protection stub. The level estimator (AdaptiveModeLevelEstimator) produces a biased estimate of the speech level. In our model, we use another module (the SaturationProtector) to compute the bias. This CL contains the estimator and a stub of the saturation protector. Bug: webrtc:7494 Change-Id: I0df736d0346063f544fa680b4cc84177ea548545 Reviewed-on: https://webrtc-review.googlesource.com/64820 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22641}
2018-03-28 09:45:29 +02:00
"agc2:adaptive_digital_unittests",
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
"agc2:biquad_filter_unittests",
Add FixedGainController and move GainController2 in APM. The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 11:54:07 +01:00
"agc2:fixed_digital_unittests",
Noise level estimation for AGC2. We put back the old noise estimator from LevelController. We add a few new unit tests. We also re-arrange the code so that it fits with how it is used in AGC2. The differences are: 1. The NoiseLevelEstimator is now fully self-contained. 2. The NoiseLevelEstimator is responsible for calling SignalClassifier and computing the signal energy. Previously the signal type and energy were used in several places. It made sense to compute the values independently of the noise calculation. 3. Re-initialization doesn't have to be done by the caller. 4. The interface is AudioFrameView instead of AudioBuffer. # Bots are green, nothing should break internal stuff NOTRY=True Bug: webrtc:7494 Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d Reviewed-on: https://webrtc-review.googlesource.com/66380 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22738}
2018-04-04 15:05:57 +02:00
"agc2:noise_estimator_unittests",
AGC2 RNN VAD: Sequence buffer The SequenceBuffer class template implements a linear buffer with a Push operation that is used to add a fixed size chunk of new samples into the buffer. Its properties are its size and the size of the chunks that are pushed. It is used to implement the pitch buffer in the RNN VAD feature extractor, for which a ring buffer would be a painful choice. Bug: webrtc:9076 Change-Id: I4767bf06d5a414dbed724a96ea4186ef013a1e30 Reviewed-on: https://webrtc-review.googlesource.com/70204 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22919}
2018-04-17 16:46:45 +02:00
"agc2/rnn_vad:unittests",
Conversational Speech tool, rtc_test target replaced with entry in modules_unittests The ConversationalSpeechTest.* unit tests are now part of modules_unittests. The rtc_test target has been replaced with an rtc_source_set one. The latter is included as dependency in audio_processing_unittests. BUG=webrtc:7218 Review-Url: https://codereview.webrtc.org/2769863005 Cr-Commit-Position: refs/heads/master@{#17360}
2017-03-23 05:17:06 -07:00
"test/conversational_speech:unittest",
Create separate build targets for utility/ in APM This clarifies the dependencies of utility/ a lot (spoiler: there are very few) and makes it easier to separate the build targets for aecm and aec2. Bug: webrtc:9488 Change-Id: If916f86e80c19d1b650d0908fbe8343ea7c47bd7 Reviewed-on: https://webrtc-review.googlesource.com/87141 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23860}
2018-07-05 16:41:55 +02:00
"utility:block_mean_calculator_unittest",
"utility:legacy_delay_estimator_unittest",
Made modules/audio_processing/vad its own target. WHAT: made a BUILD.gn with library and tests in the Audio Processing Module Voice Activity Detector directory. Updated depending code. Fixed a Clang warning. WHY: to make it possible for a target to depend on just the VAD and not the whole APM. There are other benefits: * Sometimes faster compilation. * The VAD takes up 28000 bytes of libjingle_peerconnection_so.so. Making a peerconnection shared object file without the VAD has to be done in steps. The first step is a custom target for the VAD. Hence this Cl. Change-Id: Iea0207a0b5979db26baaf46b24beaefbb1c431af BUG: webrtc:5716, webrtc:7494 Reviewed-on: https://webrtc-review.googlesource.com/47521 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21893}
2018-02-05 11:15:23 +01:00
"vad:vad_unittests",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"//testing/gtest",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace rtc::Optional with absl::optional in modules/audio processing This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_processing' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0 Reviewed-on: https://webrtc-review.googlesource.com/83982 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23653}
2018-06-19 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
]
defines = []
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",
Separate build targets for APM bit exactness tools from unittests. This places the bit exactness testing tools in audioproc_test_utils, and removes it from audio_processing_unittests. Bug: webrtc:8240 Change-Id: I6f54ea3c49c0212888c6f8a779ecc886d1d2baba Reviewed-on: https://chromium-review.googlesource.com/663545 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19815}
2017-09-13 11:00:04 +02:00
":audioproc_test_utils",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":audioproc_unittest_proto",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
Add UTC time to init event in AEC debug dump. Bug: webrtc:9616 Change-Id: I1350212f0b8835fb64427483269da96d51670c01 Reviewed-on: https://webrtc-review.googlesource.com/92620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24267}
2018-08-10 15:38:52 +02:00
"../../rtc_base:rtc_base_tests_utils",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_task_queue",
Change existing aec dump tests to use webrtc::AecDump. Currently the debug dump functionality of WebRTC (a log of all AudioProcessing operations) was tested by the following tests: 1. ApmTest.VerifyDebugDump* which configures and runs AudioProcessing from a debug dump, and verifies that the same debug dump is recorded. 2. DebugDumpTest.* which is a comprehensive test of the debug dump operations. AudioProcessing configuration is changed, and the dump is scanned for the change. 3. ApmTest::{DebugDump, DebugDumpFromFileHandle} that verify that debug dumping can be started and files written. This CL replaces the debug dump mechanism in all these tests to webrtc::AecDump. Some of the tests are adapted to the chenges of the new API to AecDump {Start,Stop}DebugRecording: the old functions signal errors when a file cannot be opened. With AecDump, the AecDumpFactory instead returns a nullptr. The CL also changes audioproc_f to use AecDump. BUG=webrtc:7404 Review-Url: https://codereview.webrtc.org/2864373002 Cr-Commit-Position: refs/heads/master@{#18605}
2017-06-15 01:55:38 -07:00
"aec_dump",
"aec_dump:aec_dump_unittests",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
]
sources += [
"audio_processing_impl_locking_unittest.cc",
"audio_processing_impl_unittest.cc",
"audio_processing_unittest.cc",
"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_estimator_unittest.cc",
"low_cut_filter_unittest.cc",
"noise_suppression_unittest.cc",
"residual_echo_detector_unittest.cc",
"rms_level_unittest.cc",
"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") {
testonly = true
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ ) Reason for revert: Try again. Original issue's description: > Revert of Refactor webrtc_perf_tests into several source_sets. (patchset #5 id:100001 of https://codereview.webrtc.org/2609403002/ ) > > Reason for revert: > Probably cause unexpected performance regression > BUG=chromium:678569 > > Original issue's description: > > Refactor webrtc_perf_tests into several source_sets. > > > > BUG=webrtc:6954 > > > > Review-Url: https://codereview.webrtc.org/2609403002 > > Cr-Commit-Position: refs/heads/master@{#15902} > > Committed: https://chromium.googlesource.com/external/webrtc/+/0b5a26a5768618e5d850aa7ea97cb348fe5302f7 > > TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2613913002 > Cr-Commit-Position: refs/heads/master@{#15916} > Committed: https://chromium.googlesource.com/external/webrtc/+/5fbcd228f0781526ce2cbafba0a1f39f481ad81b TBR=kjellander@webrtc.org,danilchap@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:678569 Review-Url: https://codereview.webrtc.org/2615873002 Cr-Commit-Position: refs/heads/master@{#15919}
2017-01-05 07:09:50 -08:00
sources = [
"audio_processing_performance_unittest.cc",
]
deps = [
":audio_processing",
":audioproc_test_utils",
Removed residual echo complexity unittest. This test produces a consistent stream of false positive alerts, and I have been unable to make it more robust, despite several attempts. It also has never managed to catch a real regression, so I think it is better to remove it. Bug: chromium:788318 Change-Id: I7e9731834f67af1ef2fa15a655e620bd64a4cfde Reviewed-on: https://webrtc-review.googlesource.com/25824 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20874}
2017-11-24 15:35:57 +01:00
"../../api:array_view",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
Removed residual echo complexity unittest. This test produces a consistent stream of false positive alerts, and I have been unable to make it more robust, despite several attempts. It also has never managed to catch a real regression, so I think it is better to remove it. Bug: chromium:788318 Change-Id: I7e9731834f67af1ef2fa15a655e620bd64a4cfde Reviewed-on: https://webrtc-review.googlesource.com/25824 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20874}
2017-11-24 15:35:57 +01:00
"../../rtc_base:rtc_base_approved",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Reland "iOS: Save perf results under Documents/perf_result.json" This will require a manual roll to downstream projects, since the //test:perf_test target was introduced. This is a reland of 10a8e7a9b5261a7e3ce19900ba3511be3b5911f8 Original change's description: > iOS: Save perf results under Documents/perf_result.json > > TBR=henrika@webrtc.org > > Bug: webrtc:7156 > Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5 > Reviewed-on: https://webrtc-review.googlesource.com/29202 > Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21244} R=henrika@webrtc.org, phoglund@webrtc.org Bug: webrtc:7156 Change-Id: I85fc7bc5fce0894af90017b71b9952b61b523424 Reviewed-on: https://webrtc-review.googlesource.com/37643 Reviewed-by: Patrik Höglund <phoglund@google.com> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21518}
2018-01-05 15:34:09 +01:00
"../../test:perf_test",
Removed residual echo complexity unittest. This test produces a consistent stream of false positive alerts, and I have been unable to make it more robust, despite several attempts. It also has never managed to catch a real regression, so I think it is better to remove it. Bug: chromium:788318 Change-Id: I7e9731834f67af1ef2fa15a655e620bd64a4cfde Reviewed-on: https://webrtc-review.googlesource.com/25824 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20874}
2017-11-24 15:35:57 +01:00
"../../test:test_support",
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
]
}
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
rtc_source_set("file_audio_generator_unittests") {
testonly = true
sources = [
"audio_generator/file_audio_generator_unittest.cc",
]
deps = [
":audio_generator_factory",
":audio_processing",
":file_audio_generator",
"../../rtc_base:rtc_base_approved",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Add stub draft of audio generator to APM This provides the empty shell of an AudioGenerator class. It is intended to be used for debugging purposes and can be inserted into the APM much like an AecDump. It allows for playing out diagnostic audio unaffected by codecs and network jitter, while still capturing API interaction like in a normal call. NOTRY=True Bug: webrtc:8882 Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7 Reviewed-on: https://webrtc-review.googlesource.com/53300 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22282}
2018-03-02 16:03:21 +01:00
"../../test:test_support",
]
}
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
rtc_source_set("analog_mic_simulation") {
sources = [
"test/fake_recording_device.cc",
"test/fake_recording_device.h",
]
deps = [
"../../api:array_view",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
"../../common_audio:common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
"../../rtc_base:rtc_base_approved",
Add AGC1-compliant fake recording device. The AGC submodule of APM changes analog gain. These gain changes are typically ignored by the test tool audioproc_f. There is an option of the test tool to take action on the gain changes. It's the '--simulate_mic_gain' option. The option converts the analog gain to a digital gain. The digital gain is applied to the capture stream. This change adds a new simulated microphone kind. The new microphone has a gain curve defined by modules/audio_processing/agc/gain_map_internal.h. That gain curve defines how AGC1 expects a microphone to behave. Bug: webrtc:7494 Change-Id: Ifb3f54a8c6f8c001a711fa977f39f32413069780 Reviewed-on: https://webrtc-review.googlesource.com/86128 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23801}
2018-07-02 11:37:47 +02:00
"../../rtc_base:safe_minmax",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"agc:gain_map",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace rtc::Optional with absl::optional in modules/audio processing This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_processing' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0 Reviewed-on: https://webrtc-review.googlesource.com/83982 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23653}
2018-06-19 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
]
}
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
if (rtc_enable_protobuf) {
Restructure the audioproc_f tool into a library with a thin executable wrapper. This refactoring makes it easier to experiment with injectable components. Bug: webrtc:8732 Change-Id: I2cd2a8ff80516a76aec814af02b61778915f2217 Reviewed-on: https://webrtc-review.googlesource.com/60863 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22372}
2018-03-09 17:33:04 +01:00
rtc_source_set("audioproc_f_impl") {
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
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",
Moved audioproc_f interface into api directory. The interface of the audioproc_f tool should be located in the api/ directory, so it becomes visible to the outside world. Bug: webrtc:8732 Change-Id: Ia7475883aeb0e1f7a6afa5e791204b38dc53a8b8 Reviewed-on: https://webrtc-review.googlesource.com/61801 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22449}
2018-03-15 12:22:52 +01:00
"test/audioproc_float_impl.cc",
"test/audioproc_float_impl.h",
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
"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 = [
audioproc_f with simulated mic analog gain The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic. The simulation is done by applying digital gain to the input samples. This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal. This CL has been ported from https://codereview.webrtc.org/2834643002/. Bug: webrtc:7494 Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48 Reviewed-on: https://webrtc-review.googlesource.com/2685 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19992}
2017-09-27 11:51:10 +02:00
":analog_mic_simulation",
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
":audio_processing",
":audioproc_debug_proto",
":audioproc_protobuf_utils",
":audioproc_test_utils",
Move EchoCanceller3Factory to api/auido The AEC3 factory is now part of the WebRTC API. Bug: webrtc:8844 Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e Reviewed-on: https://webrtc-review.googlesource.com/57141 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22204}
2018-02-27 13:58:45 +01:00
"../../api/audio:aec3_factory",
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
"../../common_audio:common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
Extend the audioproc_f input parameters to match what is supported by AEC3 This CL extends the options for the audioproc_f tool to match the options for AEC3. Bug: webrtc:8671 Change-Id: I39972eae33dba461b94118ec47a8560eb9cfe5a6 Reviewed-on: https://webrtc-review.googlesource.com/43120 Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22344}
2018-03-08 07:47:14 +01:00
"../../rtc_base:rtc_json",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_task_queue",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../../rtc_base:stringutils",
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
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
Change existing aec dump tests to use webrtc::AecDump. Currently the debug dump functionality of WebRTC (a log of all AudioProcessing operations) was tested by the following tests: 1. ApmTest.VerifyDebugDump* which configures and runs AudioProcessing from a debug dump, and verifies that the same debug dump is recorded. 2. DebugDumpTest.* which is a comprehensive test of the debug dump operations. AudioProcessing configuration is changed, and the dump is scanned for the change. 3. ApmTest::{DebugDump, DebugDumpFromFileHandle} that verify that debug dumping can be started and files written. This CL replaces the debug dump mechanism in all these tests to webrtc::AecDump. Some of the tests are adapted to the chenges of the new API to AecDump {Start,Stop}DebugRecording: the old functions signal errors when a file cannot be opened. With AecDump, the AecDumpFactory instead returns a nullptr. The CL also changes audioproc_f to use AecDump. BUG=webrtc:7404 Review-Url: https://codereview.webrtc.org/2864373002 Cr-Commit-Position: refs/heads/master@{#18605}
2017-06-15 01:55:38 -07:00
"aec_dump",
"aec_dump:aec_dump_impl",
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
"//testing/gtest",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace rtc::Optional with absl::optional in modules/audio processing This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_processing' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0 Reviewed-on: https://webrtc-review.googlesource.com/83982 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23653}
2018-06-19 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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
]
Restructure the audioproc_f tool into a library with a thin executable wrapper. This refactoring makes it easier to experiment with injectable components. Bug: webrtc:8732 Change-Id: I2cd2a8ff80516a76aec814af02b61778915f2217 Reviewed-on: https://webrtc-review.googlesource.com/60863 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22372}
2018-03-09 17:33:04 +01:00
} # audioproc_f_impl
rtc_executable("audioproc_f") {
testonly = true
sources = [
"test/audioproc_float_main.cc",
]
deps = [
":audio_processing",
Moved audioproc_f interface into api directory. The interface of the audioproc_f tool should be located in the api/ directory, so it becomes visible to the outside world. Bug: webrtc:8732 Change-Id: Ia7475883aeb0e1f7a6afa5e791204b38dc53a8b8 Reviewed-on: https://webrtc-review.googlesource.com/61801 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22449}
2018-03-15 12:22:52 +01:00
"../../api:audioproc_f_api",
Restructure the audioproc_f tool into a library with a thin executable wrapper. This refactoring makes it easier to experiment with injectable components. Bug: webrtc:8732 Change-Id: I2cd2a8ff80516a76aec814af02b61778915f2217 Reviewed-on: https://webrtc-review.googlesource.com/60863 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22372}
2018-03-09 17:33:04 +01:00
"../../rtc_base:rtc_base_approved",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Restructure the audioproc_f tool into a library with a thin executable wrapper. This refactoring makes it easier to experiment with injectable components. Bug: webrtc:8732 Change-Id: I2cd2a8ff80516a76aec814af02b61778915f2217 Reviewed-on: https://webrtc-review.googlesource.com/60863 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22372}
2018-03-09 17:33:04 +01:00
]
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
} # 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") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
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",
Separate build targets for APM bit exactness tools from unittests. This places the bit exactness testing tools in audioproc_test_utils, and removes it from audio_processing_unittests. Bug: webrtc:8240 Change-Id: I6f54ea3c49c0212888c6f8a779ecc886d1d2baba Reviewed-on: https://chromium-review.googlesource.com/663545 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19815}
2017-09-13 11:00:04 +02:00
"test/bitexactness_tools.cc",
"test/bitexactness_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",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
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
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Separate build targets for APM bit exactness tools from unittests. This places the bit exactness testing tools in audioproc_test_utils, and removes it from audio_processing_unittests. Bug: webrtc:8240 Change-Id: I6f54ea3c49c0212888c6f8a779ecc886d1d2baba Reviewed-on: https://chromium-review.googlesource.com/663545 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19815}
2017-09-13 11:00:04 +02:00
"../../test:test_support",
Added experiment to improve handling of frame length changes in NetEq. The field trial effects two things: after a frame length change the IAT histogram is scaled to prevent an immediate change in target buffer level. Also, the peak history in the delay peak detector is cleared, because the size of the peaks is stored in number of packets (which will be incorrect after a frame length change). Bug: webrtc:8381 Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8 Reviewed-on: https://webrtc-review.googlesource.com/8101 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20284}
2017-10-13 12:37:27 +02:00
"../audio_coding:neteq_input_audio_tools",
Separate build targets for APM bit exactness tools from unittests. This places the bit exactness testing tools in audioproc_test_utils, and removes it from audio_processing_unittests. Bug: webrtc:8240 Change-Id: I6f54ea3c49c0212888c6f8a779ecc886d1d2baba Reviewed-on: https://chromium-review.googlesource.com/663545 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19815}
2017-09-13 11:00:04 +02:00
"//testing/gtest",
Replace rtc::Optional with absl::optional in modules/audio processing This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'modules/audio_processing' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0 Reviewed-on: https://webrtc-review.googlesource.com/83982 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23653}
2018-06-19 10:50:11 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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",
Replace remaining gflags usages with rtc_base/flags Continued from https://codereview.webrtc.org/2995363002 BUG=webrtc:7644 Review-Url: https://codereview.webrtc.org/3005483002 Cr-Commit-Position: refs/heads/master@{#19624}
2017-08-31 03:21:39 -07:00
"../../rtc_base:rtc_base_approved",
Move file_wrapper.h to rtc_base/system/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 Change-Id: I440974da4d347b09ff042478720d7983056b62b9 Reviewed-on: https://webrtc-review.googlesource.com/21226 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22579}
2018-03-23 10:39:34 +01:00
"../../rtc_base/system:file_wrapper",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Reland: Fix all circular deps in audio_processing (but one). Arguably we should add a few more targets, for instance a utility target, but I tried to create as few targets as possible here based on the current usage. Bug: webrtc:6828 Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8 Reviewed-on: https://webrtc-review.googlesource.com/28301 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21199}
2017-12-11 10:28:16 +01:00
"../../system_wrappers:metrics_default",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
GN 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",
Break out Agc code from audio_processing. Splits 'modules/audio_processing:audio_processing' target. The files in modules/audio_processing/agc now are in targets in that folder. Reason for doing this was to include modules/audio_processing/agc/agc.h from another target in the dependent CL https://webrtc-review.googlesource.com/c/src/+/86603 This could help reducing the binary size in the future. Bug: webrtc:7494 Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887 Reviewed-on: https://webrtc-review.googlesource.com/87422 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23873}
2018-07-06 14:54:30 +02:00
"agc:level_estimation",
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
"//testing/gtest",
]
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_executable("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",
Move file_wrapper.h to rtc_base/system/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 Change-Id: I440974da4d347b09ff042478720d7983056b62b9 Reviewed-on: https://webrtc-review.googlesource.com/21226 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22579}
2018-03-23 10:39:34 +01:00
"../../rtc_base/system:file_wrapper",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
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",
]
}
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",
]
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
proto_out_dir = "modules/audio_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",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
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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