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# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
import("//build/config/arm.gni")
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")
if (!build_with_mozilla) {
import("//third_party/protobuf/proto_library.gni")
}
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
}
config("apm_debug_dump") {
if (apm_debug_dump) {
defines = [ "WEBRTC_APM_DEBUG_DUMP=1" ]
} else {
defines = [ "WEBRTC_APM_DEBUG_DUMP=0" ]
}
}
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 = [ "*" ]
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 = [
"aec/aec_resampler.cc",
"aec/aec_resampler.h",
"aec/echo_cancellation.cc",
"aec/echo_cancellation.h",
"agc/agc.cc",
"agc/agc.h",
"agc/agc_manager_direct.cc",
"agc/agc_manager_direct.h",
"agc/gain_map_internal.h",
"agc/loudness_histogram.cc",
"agc/loudness_histogram.h",
"agc/utility.cc",
"agc/utility.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"audio_buffer.cc",
"audio_buffer.h",
"audio_processing_impl.cc",
"audio_processing_impl.h",
"beamformer/array_util.cc",
"beamformer/array_util.h",
"beamformer/complex_matrix.h",
"beamformer/covariance_matrix_generator.cc",
"beamformer/covariance_matrix_generator.h",
"beamformer/matrix.h",
"beamformer/nonlinear_beamformer.cc",
"beamformer/nonlinear_beamformer.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
"common.h",
"echo_cancellation_impl.cc",
"echo_cancellation_impl.h",
"echo_control_mobile_impl.cc",
"echo_control_mobile_impl.h",
"echo_detector/circular_buffer.cc",
"echo_detector/circular_buffer.h",
"echo_detector/mean_variance_estimator.cc",
"echo_detector/mean_variance_estimator.h",
"echo_detector/moving_max.cc",
"echo_detector/moving_max.h",
"echo_detector/normalized_covariance_estimator.cc",
"echo_detector/normalized_covariance_estimator.h",
"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",
"gain_controller2.cc",
"gain_controller2.h",
"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",
"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",
"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",
"render_queue_item_verifier.h",
"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",
"splitting_filter.cc",
"splitting_filter.h",
"three_band_filter_bank.cc",
"three_band_filter_bank.h",
"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 = []
deps = [
":aec_core",
":aec_dump_interface",
":apm_logging",
":audio_frame_view",
":audio_generator_interface",
":audio_processing_c",
":audio_processing_statistics",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../api:array_view",
"../../api:optional",
"../../api/audio:aec3_config",
"../../api/audio:echo_control",
"../../audio/utility:audio_frame_operations",
"../../common_audio:common_audio_c",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:gtest_prod",
"../../rtc_base:protobuf_utils",
"../../rtc_base:safe_minmax",
"../../rtc_base:sanitizer",
"../../system_wrappers:cpu_features_api",
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
"agc2",
"vad",
]
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" ]
}
if (rtc_enable_protobuf) {
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
deps += [ ":audioproc_debug_proto" ]
}
if (rtc_enable_intelligibility_enhancer) {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
sources += [
"intelligibility/intelligibility_enhancer.cc",
"intelligibility/intelligibility_enhancer.h",
"intelligibility/intelligibility_utils.cc",
"intelligibility/intelligibility_utils.h",
]
} else {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
}
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" ]
}
# TODO(jschuh): Bug 1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
deps += [
"../../common_audio",
"../../common_audio:fir_filter",
"../../common_audio:fir_filter_factory",
"../../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",
]
}
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 = [ "*" ]
sources = [
"include/audio_processing_statistics.cc",
"include/audio_processing_statistics.h",
]
deps = [
"../../api:optional",
]
}
rtc_source_set("audio_frame_view") {
sources = [
"include/audio_frame_view.h",
]
deps = [
"../../api:array_view",
]
}
rtc_source_set("aec_dump_interface") {
sources = [
"include/aec_dump.cc",
"include/aec_dump.h",
]
deps = [
":audio_frame_view",
"../../api:array_view",
"../../rtc_base:rtc_base_approved",
]
}
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",
]
}
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",
]
}
rtc_source_set("audio_processing_c") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
sources = [
"agc/legacy/analog_agc.c",
"agc/legacy/analog_agc.h",
"agc/legacy/digital_agc.c",
"agc/legacy/digital_agc.h",
"agc/legacy/gain_control.h",
]
if (rtc_prefer_fixed_point) {
sources += [
"ns/noise_suppression_x.c",
"ns/noise_suppression_x.h",
"ns/nsx_core.c",
"ns/nsx_core.h",
"ns/nsx_defines.h",
]
if (current_cpu == "mipsel") {
sources += [ "ns/nsx_core_mips.c" ]
} else {
sources += [ "ns/nsx_core_c.c" ]
}
} else {
sources += [
"ns/defines.h",
"ns/noise_suppression.c",
"ns/noise_suppression.h",
"ns/ns_core.c",
"ns/ns_core.h",
"ns/windows_private.h",
]
}
deps = [
"../..:typedefs",
"../..:webrtc_common",
"../../common_audio",
"../../common_audio:common_audio_c",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:cpu_features_api",
]
if (rtc_build_with_neon) {
sources += [ "ns/nsx_core_neon.c" ]
if (current_cpu != "arm64") {
# Enable compilation for the NEON instruction set. This is needed
# since //build/config/arm.gni only enables NEON for iOS, not Android.
# This provides the same functionality as webrtc/build/arm_neon.gypi.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
}
# Disable LTO on NEON targets due to compiler bug.
# TODO(fdegans): Enable this. See crbug.com/408997.
if (rtc_use_lto) {
cflags -= [
"-flto",
"-ffat-lto-objects",
]
}
}
}
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") {
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
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
}
}
rtc_source_set("apm_logging") {
configs += [ ":apm_debug_dump" ]
sources = [
"logging/apm_data_dumper.cc",
"logging/apm_data_dumper.h",
]
deps = [
"../../api:array_view",
"../../common_audio:common_audio",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:stringutils",
]
defines = []
}
rtc_source_set("aec_core") {
configs += [ ":apm_debug_dump" ]
sources = [
"aec/aec_common.h",
"aec/aec_core.cc",
"aec/aec_core.h",
"aec/aec_core_optimized_methods.h",
"aecm/aecm_core.cc",
"aecm/aecm_core.h",
"aecm/aecm_defines.h",
"aecm/echo_control_mobile.cc",
"aecm/echo_control_mobile.h",
"utility/block_mean_calculator.cc",
"utility/block_mean_calculator.h",
"utility/delay_estimator.cc",
"utility/delay_estimator.h",
"utility/delay_estimator_internal.h",
"utility/delay_estimator_wrapper.cc",
"utility/delay_estimator_wrapper.h",
"utility/ooura_fft.cc",
"utility/ooura_fft.h",
"utility/ooura_fft_tables_common.h",
]
deps = [
":apm_logging",
":audio_processing_statistics",
"../..:typedefs",
"../..:webrtc_common",
"../../common_audio:common_audio",
"../../common_audio:common_audio_c",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:sanitizer",
"../../system_wrappers:cpu_features_api",
"../../system_wrappers:metrics_api",
]
cflags = []
defines = []
if (current_cpu == "x86" || current_cpu == "x64") {
sources += [
"aec/aec_core_sse2.cc",
"utility/ooura_fft_sse2.cc",
"utility/ooura_fft_tables_neon_sse2.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
if (is_posix) {
cflags += [ "-msse2" ]
This CL introduces a new data logging functionality to use for the APM. It allows simple and rapid additions of exploratory data logpoints to use during bug investigations and module performance analysis. The new data logging functionality is also in this CL used to replace the existing data logging functionality present in the AEC. Additional information: As there was an issue with that the build flag for activating this feature was not present in all compilation units that included the feature additional changes were needed. A summary of the changes are -The build files were modified to ensure that the logging build flag always is set to either 0 or 1 for compilation units that include the feature. -Build-time checks in the appropriate places were added to ensure that the above is fulfilled. -The build object was added dynamically to the AEC state as a pointer to ensure that the size of that state is not dependent on whether the logging build flag is set or not. -The constructor of the AEC class needed to be modified in order to construct the logging object. For this a destructor was also needed. -An unused method without any declaration was removed in order to avoid any issues with the logging flag being set to 0 or 1. This CL will be immediately followed with an upcoming CL that replaces the logging in echo_cancellation.cc with the new functionality which will ensure that the logging flag is only used in one place within WebRTC, which in turn will fully ensure that all compilation units that uses the feature also have the flag properly set. BUG=webrtc:5201, webrtc:5298 Review-Url: https://codereview.webrtc.org/1877713002 Cr-Commit-Position: refs/heads/master@{#12607}
2016-05-03 07:01:18 -07:00
}
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
}
if (rtc_build_with_neon) {
sources += [
"aec/aec_core_neon.cc",
"aecm/aecm_core_neon.cc",
"utility/ooura_fft_neon.cc",
"utility/ooura_fft_tables_neon_sse2.h",
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
]
if (current_cpu != "arm64") {
# Enable compilation for the NEON instruction set. This is needed
# since //build/config/arm.gni only enables NEON for iOS, not Android.
# This provides the same functionality as webrtc/build/arm_neon.gypi.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags += [ "-mfpu=neon" ]
}
# Disable LTO on NEON targets due to compiler bug.
# TODO(fdegans): Enable this. See crbug.com/408997.
if (rtc_use_lto) {
GN: Implement voice engine, common audio, audio coding and audio processing NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
cflags -= [
"-flto",
"-ffat-lto-objects",
]
}
deps += [ "../../common_audio" ]
}
This CL introduces a new data logging functionality to use for the APM. It allows simple and rapid additions of exploratory data logpoints to use during bug investigations and module performance analysis. The new data logging functionality is also in this CL used to replace the existing data logging functionality present in the AEC. Additional information: As there was an issue with that the build flag for activating this feature was not present in all compilation units that included the feature additional changes were needed. A summary of the changes are -The build files were modified to ensure that the logging build flag always is set to either 0 or 1 for compilation units that include the feature. -Build-time checks in the appropriate places were added to ensure that the above is fulfilled. -The build object was added dynamically to the AEC state as a pointer to ensure that the size of that state is not dependent on whether the logging build flag is set or not. -The constructor of the AEC class needed to be modified in order to construct the logging object. For this a destructor was also needed. -An unused method without any declaration was removed in order to avoid any issues with the logging flag being set to 0 or 1. This CL will be immediately followed with an upcoming CL that replaces the logging in echo_cancellation.cc with the new functionality which will ensure that the logging flag is only used in one place within WebRTC, which in turn will fully ensure that all compilation units that uses the feature also have the flag properly set. BUG=webrtc:5201, webrtc:5298 Review-Url: https://codereview.webrtc.org/1877713002 Cr-Commit-Position: refs/heads/master@{#12607}
2016-05-03 07:01:18 -07:00
if (current_cpu == "mipsel") {
sources += [ "aecm/aecm_core_mips.cc" ]
if (mips_float_abi == "hard") {
sources += [
"aec/aec_core_mips.cc",
"utility/ooura_fft_mips.cc",
]
This CL introduces a new data logging functionality to use for the APM. It allows simple and rapid additions of exploratory data logpoints to use during bug investigations and module performance analysis. The new data logging functionality is also in this CL used to replace the existing data logging functionality present in the AEC. Additional information: As there was an issue with that the build flag for activating this feature was not present in all compilation units that included the feature additional changes were needed. A summary of the changes are -The build files were modified to ensure that the logging build flag always is set to either 0 or 1 for compilation units that include the feature. -Build-time checks in the appropriate places were added to ensure that the above is fulfilled. -The build object was added dynamically to the AEC state as a pointer to ensure that the size of that state is not dependent on whether the logging build flag is set or not. -The constructor of the AEC class needed to be modified in order to construct the logging object. For this a destructor was also needed. -An unused method without any declaration was removed in order to avoid any issues with the logging flag being set to 0 or 1. This CL will be immediately followed with an upcoming CL that replaces the logging in echo_cancellation.cc with the new functionality which will ensure that the logging flag is only used in one place within WebRTC, which in turn will fully ensure that all compilation units that uses the feature also have the flag properly set. BUG=webrtc:5201, webrtc:5298 Review-Url: https://codereview.webrtc.org/1877713002 Cr-Commit-Position: refs/heads/master@{#12607}
2016-05-03 07:01:18 -07:00
}
} else {
sources += [ "aecm/aecm_core_c.cc" ]
}
# TODO(jschuh): Bug 1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
}
if (rtc_include_tests) {
rtc_source_set("mocks") {
testonly = true
sources = [
"include/mock_audio_processing.h",
]
deps = [
":aec_dump_interface",
":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
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",
":nonlinear_beamformer_test",
":transient_suppression_test",
]
if (rtc_enable_intelligibility_enhancer) {
deps += [ ":intelligibility_proc" ]
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
}
if (rtc_enable_protobuf) {
deps += [
":audioproc_f",
":audioproc_unittest_proto",
"aec_dump:aec_dump_unittests",
"test/conversational_speech",
"test/py_quality_assessment",
]
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
}
}
rtc_source_set("audio_processing_unittests") {
testonly = true
configs += [ ":apm_debug_dump" ]
sources = [
"aec/echo_cancellation_unittest.cc",
"aec/system_delay_unittest.cc",
"agc/agc_manager_direct_unittest.cc",
"agc/loudness_histogram_unittest.cc",
"agc/mock_agc.h",
"audio_buffer_unittest.cc",
"audio_frame_view_unittest.cc",
"beamformer/array_util_unittest.cc",
"beamformer/complex_matrix_unittest.cc",
"beamformer/covariance_matrix_generator_unittest.cc",
"beamformer/matrix_test_helpers.h",
"beamformer/matrix_unittest.cc",
"beamformer/mock_nonlinear_beamformer.h",
"config_unittest.cc",
"echo_cancellation_impl_unittest.cc",
"gain_controller2_unittest.cc",
"splitting_filter_unittest.cc",
"test/fake_recording_device_unittest.cc",
"transient/dyadic_decimator_unittest.cc",
"transient/file_utils.cc",
"transient/file_utils.h",
"transient/file_utils_unittest.cc",
"transient/moving_moments_unittest.cc",
"transient/transient_detector_unittest.cc",
"transient/transient_suppressor_unittest.cc",
"transient/wpd_node_unittest.cc",
"transient/wpd_tree_unittest.cc",
"utility/block_mean_calculator_unittest.cc",
"utility/delay_estimator_unittest.cc",
]
deps = [
":aec_core",
":analog_mic_simulation",
":apm_logging",
":audio_frame_view",
":audio_processing",
":audioproc_test_utils",
":file_audio_generator_unittests",
":mocks",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../api:array_view",
"../../api:optional",
"../../api/audio:aec3_config",
"../../api/audio:aec3_factory",
"../../common_audio:common_audio",
"../../common_audio:common_audio_c",
"../../rtc_base:checks",
"../../rtc_base:gtest_prod",
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:safe_minmax",
"../../system_wrappers",
"../../system_wrappers:cpu_features_api",
"../../test:test_support",
"../audio_coding:neteq_input_audio_tools",
"aec_dump:mock_aec_dump_unittests",
"agc2:fixed_digital_unittests",
"test/conversational_speech:unittest",
"vad:vad_unittests",
"//testing/gtest",
]
defines = []
if (rtc_enable_intelligibility_enhancer) {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
sources += [
"intelligibility/intelligibility_enhancer_unittest.cc",
"intelligibility/intelligibility_utils_unittest.cc",
]
} else {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
}
if (rtc_prefer_fixed_point) {
defines += [ "WEBRTC_AUDIOPROC_FIXED_PROFILE" ]
} else {
defines += [ "WEBRTC_AUDIOPROC_FLOAT_PROFILE" ]
}
if (rtc_enable_protobuf) {
defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
deps += [
":audioproc_debug_proto",
":audioproc_protobuf_utils",
":audioproc_test_utils",
":audioproc_unittest_proto",
"../../rtc_base:rtc_task_queue",
"aec_dump",
"aec_dump:aec_dump_unittests",
]
sources += [
"audio_processing_impl_locking_unittest.cc",
"audio_processing_impl_unittest.cc",
"audio_processing_unittest.cc",
"beamformer/nonlinear_beamformer_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",
"test/echo_canceller_test_tools.cc",
"test/echo_canceller_test_tools.h",
"test/echo_canceller_test_tools_unittest.cc",
"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" ]
}
}
rtc_source_set("audio_processing_perf_tests") {
testonly = true
sources = [
"audio_processing_performance_unittest.cc",
]
deps = [
":audio_processing",
":audioproc_test_utils",
"../../api:array_view",
"../../modules:module_api",
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../test:perf_test",
"../../test:test_support",
]
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ ) Reason for revert: Trying to re-land after solving some related issues. There are no changes compared to the original CL. Original issue's description: > Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ ) > > Reason for revert: > I will try to reland next week because it is causing some problems. > > Original issue's description: > > To accommodate some downstream WebRTC users we need to loosen > > the coupling between our code and the //third_party/protobuf. > > > > This includes using typedefs to define strings instead of > > assuming std::string. > > > > After this refactoring it will be possible to link with other > > protobuf implementations than the current one. > > > > We moved the PRESUBMIT check to another CL [1]. The goal of this > > presubmit is to avoid the direct usage of google::protobuf outside > > of the webrtc/base/protobuf_utils.h header file. > > > > [1] - https://codereview.webrtc.org/2753823003/ > > > > BUG=webrtc:7340 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2747863003 > > Cr-Commit-Position: refs/heads/master@{#17466} > > Committed: https://chromium.googlesource.com/external/webrtc/+/16ab93b952f9e8268f2e663ffe49548e8043d5af > > TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:7340 > > Review-Url: https://codereview.webrtc.org/2786363002 > Cr-Commit-Position: refs/heads/master@{#17483} > Committed: https://chromium.googlesource.com/external/webrtc/+/d00aad5eb2fa5a7b5aeda714f7702b50cd26ee28 TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7340 NOTRY=True Review-Url: https://codereview.webrtc.org/2791963003 Cr-Commit-Position: refs/heads/master@{#17584}
2017-04-07 00:59:12 -07:00
if (rtc_enable_intelligibility_enhancer) {
defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
} else {
defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
}
}
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",
"../../test:test_support",
]
}
rtc_source_set("analog_mic_simulation") {
sources = [
"test/fake_recording_device.cc",
"test/fake_recording_device.h",
]
deps = [
"../../api:array_view",
"../../common_audio:common_audio",
"../../modules:module_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]
}
if (rtc_enable_protobuf) {
rtc_executable("audioproc_f") {
testonly = true
sources = [
"test/aec_dump_based_simulator.cc",
"test/aec_dump_based_simulator.h",
"test/audio_processing_simulator.cc",
"test/audio_processing_simulator.h",
"test/audioproc_float.cc",
"test/wav_based_simulator.cc",
"test/wav_based_simulator.h",
]
deps = [
":analog_mic_simulation",
":audio_processing",
":audioproc_debug_proto",
":audioproc_protobuf_utils",
":audioproc_test_utils",
"../../api:optional",
"../../api/audio:aec3_factory",
"../../common_audio:common_audio",
"../../rtc_base:checks",
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_json",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:stringutils",
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
"aec_dump",
"aec_dump:aec_dump_impl",
"//testing/gtest",
]
} # audioproc_f
}
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 = [ "*" ]
testonly = true
sources = [
"test/audio_buffer_tools.cc",
"test/audio_buffer_tools.h",
"test/bitexactness_tools.cc",
"test/bitexactness_tools.h",
"test/performance_timer.cc",
"test/performance_timer.h",
"test/simulator_buffers.cc",
"test/simulator_buffers.h",
"test/test_utils.cc",
"test/test_utils.h",
]
deps = [
":audio_processing",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
"../../api:array_view",
"../../api:optional",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../test:test_support",
"../audio_coding:neteq_input_audio_tools",
"//testing/gtest",
]
}
rtc_executable("transient_suppression_test") {
testonly = true
sources = [
"transient/file_utils.cc",
"transient/file_utils.h",
"transient/transient_suppression_test.cc",
]
deps = [
":audio_processing",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
"../..:typedefs",
"../..:webrtc_common",
"../../common_audio:common_audio",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../system_wrappers:metrics_default",
"../../test:test_support",
"//testing/gtest",
]
}
rtc_executable("click_annotate") {
testonly = true
sources = [
"transient/click_annotate.cc",
"transient/file_utils.cc",
"transient/file_utils.h",
]
deps = [
":audio_processing",
"../..:typedefs",
"../..:webrtc_common",
"../../system_wrappers",
"../../system_wrappers:metrics_default",
]
}
rtc_executable("nonlinear_beamformer_test") {
testonly = true
sources = [
"beamformer/nonlinear_beamformer_test.cc",
]
deps = [
":audio_processing",
":audioproc_test_utils",
"../../common_audio:common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:metrics_default",
]
}
if (rtc_enable_intelligibility_enhancer) {
rtc_executable("intelligibility_proc") {
testonly = true
sources = [
"intelligibility/test/intelligibility_proc.cc",
]
deps = [
":audio_processing",
":audioproc_test_utils",
"../../common_audio",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:metrics_default",
"../../test:test_support",
"//testing/gtest",
]
}
}
if (rtc_enable_protobuf) {
proto_library("audioproc_unittest_proto") {
sources = [
"test/unittest.proto",
]
proto_out_dir = "modules/audio_processing/test"
}
rtc_static_library("audioproc_protobuf_utils") {
sources = [
"test/protobuf_utils.cc",
"test/protobuf_utils.h",
]
deps = [
":audioproc_debug_proto",
"../..:typedefs",
"../..:webrtc_common",
"../../rtc_base:protobuf_utils",
"../../rtc_base:rtc_base_approved",
]
}
}
}