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A simple copy of the old audio mixer to a new directory. I have added build files and renamed the mixer so that it doesn't conflict with the old one. The header includes now point to this copy of the mixer. I have also fixed some of the more obvious cases of style guide non-conformance and run 'PRESUBMIT' on the old mixer. This is a first step in the creation of a new mixing module that will replace AudioConferencMixer and OutputMixer. NOTRY=True Review-Url: https://codereview.webrtc.org/2104363003 Cr-Commit-Position: refs/heads/master@{#13378}
2016-07-04 06:33:02 -07:00
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../../webrtc.gni")
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
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_mixer") {
Stop using public_deps in modules/audio_mixer. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I74c01d5a0243c96dca504b2d696092ea35c36aa3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29860 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21105}
2017-12-05 15:33:16 +01:00
deps = [
GN: New conventions, default target and refactorings Introduce a convention on categorizing GN targets: 1. Production code 2. Tests 3. Examples 4. Tools The first two have targets spread out all over the tree, while the latter are isolated to examples/ and tools/ directories. Another new convention: Each directory's BUILD.gn file shall contain a target named similar to the directory name. This target shall contain the 'most common' production code, i.e. so that most consumers of the directory can depend on only the directory (which implicitly means that target in GN). //webrtc:webrtc_tests is changed to depend on all WebRTC tests. From now on, it's necessary to add new test targets to this dependency tree in order to get them compiled. Two new group targets are created: //webrtc/modules/audio_coding:audio_coding_tests //webrtc/modules/audio_processing:audio_processing_tests to reduce the long list of tests in //webrtc:webrtc_tests. Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted to the root target, to avoid circular dependencies due to the monolithic property of these targets (a problem we've had in the past). The 'root' target at the top level is renamed to 'default', which means GN will build this target instead of _all_ generated targets (see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default). This target now depends on everything we want to build, meaning all targets now explicitly needs to be wired up from the root target in order to get build. Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build. BUG=webrtc:6440 TESTED=git cl try --clobber NOTRY=True Review-Url: https://codereview.webrtc.org/2441383002 Cr-Commit-Position: refs/heads/master@{#14821}
2016-10-28 05:44:03 -07:00
":audio_frame_manipulator",
":audio_mixer_impl",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_mixer_impl") {
Make modules/audio_mixer:audio_mixer_impl publicly visible. Follow-up CL after offline discussion with aleloi@. Bug: None Change-Id: I24623df7d821a44656336d5623a31cde4436c94f No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/c/103982 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25008}
2018-10-05 11:19:58 +02:00
visibility = [ "*" ]
A simple copy of the old audio mixer to a new directory. I have added build files and renamed the mixer so that it doesn't conflict with the old one. The header includes now point to this copy of the mixer. I have also fixed some of the more obvious cases of style guide non-conformance and run 'PRESUBMIT' on the old mixer. This is a first step in the creation of a new mixing module that will replace AudioConferencMixer and OutputMixer. NOTRY=True Review-Url: https://codereview.webrtc.org/2104363003 Cr-Commit-Position: refs/heads/master@{#13378}
2016-07-04 06:33:02 -07:00
sources = [
Removals and renamings in the new audio mixer. Removed the OutputMixer part of the new mixer and renamed the new mixer from NewAudioConferenceMixer to AudioMixer. NOTRY=True Review-Url: https://codereview.webrtc.org/2249213005 Cr-Commit-Position: refs/heads/master@{#13883}
2016-08-24 02:20:54 -07:00
"audio_mixer_impl.cc",
"audio_mixer_impl.h",
Injectable output rate calculater for AudioMixer. This CL breaks out the output sample rate calculation from webrtc::AudioMixerImpl. A new OutputRateCalculator interface is added to make the sample rate configurable. There are at least three reasons for this change: 1. The mixer will be used for an internal project, in which no resampling is done after the mixing. There the sample rate should be static. Currently, it can differ across mix iterations and depends on the number of audio sources. If there are no sources, the WebRTC mixer behavior is to produce silence at 48 kHz. 2. A planned change to WebRTC will make audio processing steps happen at constant sample rates. A configurable sample rate calculator will make the transition simpler for the mixer. 3. The current mixer design is a single large file. Behavior is not always simple to change (e.g. as in this case to mix at a constant rate), unrelated behavior can be broken, reusing the mixer in internal projects is tricky. Using DI for the sample rate calculation solves parts of these issues. Changes: The protected mixer c-tor now takes unique_ptr<OutputRateCalculator>. The current output rate calculation is moved to DefaultOutputRateCalculator. A new factory method AudioMixerImpl::CreateWithOutputRateCalculator is added. The old factory method passes the default rate calculator. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2557713006 Cr-Commit-Position: refs/heads/master@{#15472}
2016-12-08 02:37:58 -08:00
"default_output_rate_calculator.cc",
"default_output_rate_calculator.h",
Optionally disable APM limiter in AudioMixer. The APM limiter is a component for keeping the audio from clipping by smoothly reducing the amplitude of the audio samples. It can be rather expensive because of band-splitting & merging. Also, experiments indicate that it is of questionable benefit (adding several sources of human speech almost never cause clipping). To optionally disable the limiter, this CL does some refactoring on the (quite large) AudioMixerImpl. Functionality related to actual addition of frames and handling AudioFrame meta-data (sample_rate, num_channels, samples_per_channel, time_stamp, elapsed_time_ms) is broken out in a new sub-component called FrameCombiner. The FrameCombiner is initialized with a 'use_limiter' flag. To create a mixer without using the APM limiter Inside of FrameCombiner, the meta-data handling and the audio sample addition are kept divided from each other. This also fixes a few minor GN issues so that warnings do not have to be suppressed. BUG=webrtc:7167 Review-Url: https://codereview.webrtc.org/2692333002 Cr-Commit-Position: refs/heads/master@{#16742}
2017-02-21 05:06:29 -08:00
"frame_combiner.cc",
"frame_combiner.h",
Injectable output rate calculater for AudioMixer. This CL breaks out the output sample rate calculation from webrtc::AudioMixerImpl. A new OutputRateCalculator interface is added to make the sample rate configurable. There are at least three reasons for this change: 1. The mixer will be used for an internal project, in which no resampling is done after the mixing. There the sample rate should be static. Currently, it can differ across mix iterations and depends on the number of audio sources. If there are no sources, the WebRTC mixer behavior is to produce silence at 48 kHz. 2. A planned change to WebRTC will make audio processing steps happen at constant sample rates. A configurable sample rate calculator will make the transition simpler for the mixer. 3. The current mixer design is a single large file. Behavior is not always simple to change (e.g. as in this case to mix at a constant rate), unrelated behavior can be broken, reusing the mixer in internal projects is tricky. Using DI for the sample rate calculation solves parts of these issues. Changes: The protected mixer c-tor now takes unique_ptr<OutputRateCalculator>. The current output rate calculation is moved to DefaultOutputRateCalculator. A new factory method AudioMixerImpl::CreateWithOutputRateCalculator is added. The old factory method passes the default rate calculator. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2557713006 Cr-Commit-Position: refs/heads/master@{#15472}
2016-12-08 02:37:58 -08:00
"output_rate_calculator.h",
A simple copy of the old audio mixer to a new directory. I have added build files and renamed the mixer so that it doesn't conflict with the old one. The header includes now point to this copy of the mixer. I have also fixed some of the more obvious cases of style guide non-conformance and run 'PRESUBMIT' on the old mixer. This is a first step in the creation of a new mixing module that will replace AudioConferencMixer and OutputMixer. NOTRY=True Review-Url: https://codereview.webrtc.org/2104363003 Cr-Commit-Position: refs/heads/master@{#13378}
2016-07-04 06:33:02 -07:00
]
Changed mixing api and moved resampler. Removed resampler from NewAudioConferenceMixer and AudioMixer (which started as a copy of former OutputMixer). This is part of the mixer rewrite project. In particular, this is one of the steps required to have a single mixing component instead of two doing the same thing. The next planned change (which is not part of this CL) is to plug in the new mixer (NewAudioConferenceMixer) into AudioState and AudioDeviceModule. NOTRY=True Review-Url: https://codereview.webrtc.org/2221443002 Cr-Commit-Position: refs/heads/master@{#13674}
2016-08-08 10:18:58 -07:00
public = [
Split audio mixer into interface and implementation. The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules. This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC. It will also create less build dependencies when the new mixer has replaced the old one. NOTRY=True TBR=henrik.lundin@webrtc.org BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2411313003 Cr-Commit-Position: refs/heads/master@{#14705}
2016-10-20 05:06:39 -07:00
"audio_mixer_impl.h",
Optionally disable APM limiter in AudioMixer. The APM limiter is a component for keeping the audio from clipping by smoothly reducing the amplitude of the audio samples. It can be rather expensive because of band-splitting & merging. Also, experiments indicate that it is of questionable benefit (adding several sources of human speech almost never cause clipping). To optionally disable the limiter, this CL does some refactoring on the (quite large) AudioMixerImpl. Functionality related to actual addition of frames and handling AudioFrame meta-data (sample_rate, num_channels, samples_per_channel, time_stamp, elapsed_time_ms) is broken out in a new sub-component called FrameCombiner. The FrameCombiner is initialized with a 'use_limiter' flag. To create a mixer without using the APM limiter Inside of FrameCombiner, the meta-data handling and the audio sample addition are kept divided from each other. This also fixes a few minor GN issues so that warnings do not have to be suppressed. BUG=webrtc:7167 Review-Url: https://codereview.webrtc.org/2692333002 Cr-Commit-Position: refs/heads/master@{#16742}
2017-02-21 05:06:29 -08:00
"default_output_rate_calculator.h", # For creating a mixer with limiter disabled.
"frame_combiner.h",
Changed mixing api and moved resampler. Removed resampler from NewAudioConferenceMixer and AudioMixer (which started as a copy of former OutputMixer). This is part of the mixer rewrite project. In particular, this is one of the steps required to have a single mixing component instead of two doing the same thing. The next planned change (which is not part of this CL) is to plug in the new mixer (NewAudioConferenceMixer) into AudioState and AudioDeviceModule. NOTRY=True Review-Url: https://codereview.webrtc.org/2221443002 Cr-Commit-Position: refs/heads/master@{#13674}
2016-08-08 10:18:58 -07:00
]
Reland of "Choose between APM-AGC-Limiter and Apm-AGC2-fixed-gain_controller." The webrtc::AudioMixer uses a limiter component. This CL allows changes the APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial. The AGC2 limiter has a float interface. We plan to eventually switch to the AGC2 limiter. Therefore, we will now mix in de-interleaved floats. Float mixing will happen both when using the old limiter and when using the new one. After this CL the mixer will support two limiters. The limiters have different interfaces and need different processing steps. Because of that, we make (rather big) changes to the control flow in FrameCombiner. For a short while, we will mix in deinterleaved floats when using any limiter. Originally landed in https://webrtc-review.googlesource.com/c/src/+/56141/ Reverted in https://webrtc-review.googlesource.com/c/src/+/57940 because of both breaking compilation and having a severe error. The error is fixed and a test is added. The compilation issue is fixed. Bug: webrtc:8925 Change-Id: Ieba138dee9652c826459fe637ae2dccbbc06bcf0 Reviewed-on: https://webrtc-review.googlesource.com/58085 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22207}
2018-02-27 13:51:47 +01:00
configs += [ "../audio_processing:apm_debug_dump" ]
A simple copy of the old audio mixer to a new directory. I have added build files and renamed the mixer so that it doesn't conflict with the old one. The header includes now point to this copy of the mixer. I have also fixed some of the more obvious cases of style guide non-conformance and run 'PRESUBMIT' on the old mixer. This is a first step in the creation of a new mixing module that will replace AudioConferencMixer and OutputMixer. NOTRY=True Review-Url: https://codereview.webrtc.org/2104363003 Cr-Commit-Position: refs/heads/master@{#13378}
2016-07-04 06:33:02 -07:00
deps = [
Split audio mixer into interface and implementation. The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules. This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC. It will also create less build dependencies when the new mixer has replaced the old one. NOTRY=True TBR=henrik.lundin@webrtc.org BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2411313003 Cr-Commit-Position: refs/heads/master@{#14705}
2016-10-20 05:06:39 -07:00
":audio_frame_manipulator",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
Remove APM limiter in Audio Mixer. The FrameCombiner sub-module of the AudioMixer uses one of two limiters. One is an AudioProcessingModule with AGC1 enabled and configured as a limiter. The other is the limiter part of AGC2. This change removes the APM-AGC1 limiter. This requires small changes to FrameCombiner, AudioMixerImpl and tests. We also stop using the finch experiment flag. Bug: webrtc:8925 Change-Id: Id7b8349ec4720b6417b15eaf70ed1a850b6ddbed Reviewed-on: https://webrtc-review.googlesource.com/84620 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23727}
2018-06-21 12:04:05 +02:00
"../../api/audio:audio_frame_api",
BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-16 13:43:49 +01:00
"../../api/audio:audio_mixer_api",
Move functionality out from AudioFrame and into AudioFrameOperations. This CL is in preparation to move the AudioFrame into webrtc/api. The AudioFrame is a POD type used for representing 10ms of audio. It appears as a parameter and return value of interfaces being migrated to webrtc/api, in particular AudioMixer. Here, methods operator+=, operator>>=, Mute are moved into a new target webrtc/audio/utility/audio_frame_operations, and dependencies are changed to use the new versions. The old AudioFrame methods are marked deprecated. The audio frame utilities in webrtc/modules/utility:audio_frame_operations are also moved to the new location. TBR=kjellander@webrtc.org BUG=webrtc:6548 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2424173003 Cr-Commit-Position: refs/heads/master@{#15413}
2016-12-05 01:46:09 -08:00
"../../audio/utility:audio_frame_operations",
Reland of "Choose between APM-AGC-Limiter and Apm-AGC2-fixed-gain_controller." The webrtc::AudioMixer uses a limiter component. This CL allows changes the APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial. The AGC2 limiter has a float interface. We plan to eventually switch to the AGC2 limiter. Therefore, we will now mix in de-interleaved floats. Float mixing will happen both when using the old limiter and when using the new one. After this CL the mixer will support two limiters. The limiters have different interfaces and need different processing steps. Because of that, we make (rather big) changes to the control flow in FrameCombiner. For a short while, we will mix in deinterleaved floats when using any limiter. Originally landed in https://webrtc-review.googlesource.com/c/src/+/56141/ Reverted in https://webrtc-review.googlesource.com/c/src/+/57940 because of both breaking compilation and having a severe error. The error is fixed and a test is added. The compilation issue is fixed. Bug: webrtc:8925 Change-Id: Ieba138dee9652c826459fe637ae2dccbbc06bcf0 Reviewed-on: https://webrtc-review.googlesource.com/58085 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22207}
2018-02-27 13:51:47 +01:00
"../../common_audio",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
A simple copy of the old audio mixer to a new directory. I have added build files and renamed the mixer so that it doesn't conflict with the old one. The header includes now point to this copy of the mixer. I have also fixed some of the more obvious cases of style guide non-conformance and run 'PRESUBMIT' on the old mixer. This is a first step in the creation of a new mixing module that will replace AudioConferencMixer and OutputMixer. NOTRY=True Review-Url: https://codereview.webrtc.org/2104363003 Cr-Commit-Position: refs/heads/master@{#13378}
2016-07-04 06:33:02 -07:00
"../../system_wrappers",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:metrics",
Move functionality out from AudioFrame and into AudioFrameOperations. This CL is in preparation to move the AudioFrame into webrtc/api. The AudioFrame is a POD type used for representing 10ms of audio. It appears as a parameter and return value of interfaces being migrated to webrtc/api, in particular AudioMixer. Here, methods operator+=, operator>>=, Mute are moved into a new target webrtc/audio/utility/audio_frame_operations, and dependencies are changed to use the new versions. The old AudioFrame methods are marked deprecated. The audio frame utilities in webrtc/modules/utility:audio_frame_operations are also moved to the new location. TBR=kjellander@webrtc.org BUG=webrtc:6548 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2424173003 Cr-Commit-Position: refs/heads/master@{#15413}
2016-12-05 01:46:09 -08:00
"../audio_processing",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../audio_processing:api",
Reland of "Choose between APM-AGC-Limiter and Apm-AGC2-fixed-gain_controller." The webrtc::AudioMixer uses a limiter component. This CL allows changes the APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial. The AGC2 limiter has a float interface. We plan to eventually switch to the AGC2 limiter. Therefore, we will now mix in de-interleaved floats. Float mixing will happen both when using the old limiter and when using the new one. After this CL the mixer will support two limiters. The limiters have different interfaces and need different processing steps. Because of that, we make (rather big) changes to the control flow in FrameCombiner. For a short while, we will mix in deinterleaved floats when using any limiter. Originally landed in https://webrtc-review.googlesource.com/c/src/+/56141/ Reverted in https://webrtc-review.googlesource.com/c/src/+/57940 because of both breaking compilation and having a severe error. The error is fixed and a test is added. The compilation issue is fixed. Bug: webrtc:8925 Change-Id: Ieba138dee9652c826459fe637ae2dccbbc06bcf0 Reviewed-on: https://webrtc-review.googlesource.com/58085 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22207}
2018-02-27 13:51:47 +01:00
"../audio_processing:apm_logging",
"../audio_processing:audio_frame_view",
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
"../audio_processing/agc2:fixed_digital",
Split audio mixer into interface and implementation. The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules. This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC. It will also create less build dependencies when the new mixer has replaced the old one. NOTRY=True TBR=henrik.lundin@webrtc.org BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2411313003 Cr-Commit-Position: refs/heads/master@{#14705}
2016-10-20 05:06:39 -07:00
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_frame_manipulator") {
Split audio mixer into interface and implementation. The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules. This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC. It will also create less build dependencies when the new mixer has replaced the old one. NOTRY=True TBR=henrik.lundin@webrtc.org BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2411313003 Cr-Commit-Position: refs/heads/master@{#14705}
2016-10-20 05:06:39 -07:00
visibility = [
":*",
"../../modules:*",
]
sources = [
"audio_frame_manipulator.cc",
"audio_frame_manipulator.h",
]
deps = [
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
Stop using public_deps in audio/utility. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: Ifb8df25ccb0358abcf92499a87b497cee2ab81b0 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29103 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21086}
2017-12-05 08:02:53 +01:00
"../../audio/utility:audio_frame_operations",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
A simple copy of the old audio mixer to a new directory. I have added build files and renamed the mixer so that it doesn't conflict with the old one. The header includes now point to this copy of the mixer. I have also fixed some of the more obvious cases of style guide non-conformance and run 'PRESUBMIT' on the old mixer. This is a first step in the creation of a new mixing module that will replace AudioConferencMixer and OutputMixer. NOTRY=True Review-Url: https://codereview.webrtc.org/2104363003 Cr-Commit-Position: refs/heads/master@{#13378}
2016-07-04 06:33:02 -07:00
]
}
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
if (rtc_include_tests) {
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_mixer_test_utils") {
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
testonly = true
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
sources = [
Send data from mixer to APM limiter more often. Before this change, the APM limiter used in FrameCombiner (a sub-component of AudioMixer) only gets to process the data when the number of non-muted streams is >1. If this number varies between <=1 and >1, the limiter's view of the data will have gaps during the periods with <= 1 active stream. This leads to discontinuities in the applied gain. These discontinuities cause clicks in the output audio. This change activates APM limiter processing based on the number of audio streams, independently of their mutedness status. BUG=chromium:695993 Review-Url: https://codereview.webrtc.org/2776113002 Cr-Commit-Position: refs/heads/master@{#17442}
2017-03-29 04:25:16 -07:00
"gain_change_calculator.cc",
"gain_change_calculator.h",
"sine_wave_generator.cc",
"sine_wave_generator.h",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
]
Reland of "Choose between APM-AGC-Limiter and Apm-AGC2-fixed-gain_controller." The webrtc::AudioMixer uses a limiter component. This CL allows changes the APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial. The AGC2 limiter has a float interface. We plan to eventually switch to the AGC2 limiter. Therefore, we will now mix in de-interleaved floats. Float mixing will happen both when using the old limiter and when using the new one. After this CL the mixer will support two limiters. The limiters have different interfaces and need different processing steps. Because of that, we make (rather big) changes to the control flow in FrameCombiner. For a short while, we will mix in deinterleaved floats when using any limiter. Originally landed in https://webrtc-review.googlesource.com/c/src/+/56141/ Reverted in https://webrtc-review.googlesource.com/c/src/+/57940 because of both breaking compilation and having a severe error. The error is fixed and a test is added. The compilation issue is fixed. Bug: webrtc:8925 Change-Id: Ieba138dee9652c826459fe637ae2dccbbc06bcf0 Reviewed-on: https://webrtc-review.googlesource.com/58085 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22207}
2018-02-27 13:51:47 +01:00
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
deps = [
":audio_frame_manipulator",
":audio_mixer_impl",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../../api/audio:audio_frame_api",
[getStats] Implement "media-source" audio levels, fixing Chrome bug. Implements RTCAudioSourceStats members: - audioLevel - totalAudioEnergy - totalSamplesDuration In this CL description these are collectively referred to as the audio levels. The audio levels are removed from sending "track" stats (in Chrome, these are now reported as undefined instead of 0). Background: For sending tracks, audio levels were always reported as 0 in Chrome (https://crbug.com/736403), while audio levels were correctly reported for receiving tracks. This problem affected the standard getStats() but not the legacy getStats(), blocking some people from migrating. This was likely not a problem in native third_party/webrtc code because the delivery of audio frames from device to send-stream uses a different code path outside of chromium. A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the send-side audio levels to the RTCAudioSourceStats, while keeping the receive-side audio levels on the "track" stats. This allows an implementation to report the audio levels even if samples are not sent onto the network (such as if an ICE connection has not been established yet), reflecting some of the current implementation. Changes: 1. Audio levels are added to RTCAudioSourceStats. Send-side audio "track" stats are left undefined. Receive-side audio "track" stats are not changed in this CL and continue to work. 2. Audio level computation is moved from the AudioState and AudioTransportImpl to the AudioSendStream. This is because a) the AudioTransportImpl::RecordedDataIsAvailable() code path is not exercised in chromium, and b) audio levels should, per-spec, not be calculated on a per-call basis, for which the AudioState is defined. 3. The audio level computation is now performed in AudioSendStream::SendAudioData(), a code path used by both native and chromium code. 4. Comments are added to document behavior of existing code, such as AudioLevel and AudioSendStream::SendAudioData(). Note: In this CL, just like before this CL, audio level is only calculated after an AudioSendStream has been created. This means that before an O/A negotiation, audio levels are unavailable. According to spec, if we have an audio source, we should have audio levels. An immediate solution to this would have been to calculate the audio level at pc/rtp_sender.cc. The problem is that the LocalAudioSinkAdapter::OnData() code path, while exercised in chromium, is not exercised in native code. The issue of calculating audio levels on a per-source bases rather than on a per-send stream basis is left to https://crbug.com/webrtc/10771, an existing "media-source" bug. This CL can be verified manually in Chrome at: https://codepen.io/anon/pen/vqRGyq Bug: chromium:736403, webrtc:10771 Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-03 17:11:10 +02:00
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("audio_mixer_unittests") {
[getStats] Implement "media-source" audio levels, fixing Chrome bug. Implements RTCAudioSourceStats members: - audioLevel - totalAudioEnergy - totalSamplesDuration In this CL description these are collectively referred to as the audio levels. The audio levels are removed from sending "track" stats (in Chrome, these are now reported as undefined instead of 0). Background: For sending tracks, audio levels were always reported as 0 in Chrome (https://crbug.com/736403), while audio levels were correctly reported for receiving tracks. This problem affected the standard getStats() but not the legacy getStats(), blocking some people from migrating. This was likely not a problem in native third_party/webrtc code because the delivery of audio frames from device to send-stream uses a different code path outside of chromium. A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the send-side audio levels to the RTCAudioSourceStats, while keeping the receive-side audio levels on the "track" stats. This allows an implementation to report the audio levels even if samples are not sent onto the network (such as if an ICE connection has not been established yet), reflecting some of the current implementation. Changes: 1. Audio levels are added to RTCAudioSourceStats. Send-side audio "track" stats are left undefined. Receive-side audio "track" stats are not changed in this CL and continue to work. 2. Audio level computation is moved from the AudioState and AudioTransportImpl to the AudioSendStream. This is because a) the AudioTransportImpl::RecordedDataIsAvailable() code path is not exercised in chromium, and b) audio levels should, per-spec, not be calculated on a per-call basis, for which the AudioState is defined. 3. The audio level computation is now performed in AudioSendStream::SendAudioData(), a code path used by both native and chromium code. 4. Comments are added to document behavior of existing code, such as AudioLevel and AudioSendStream::SendAudioData(). Note: In this CL, just like before this CL, audio level is only calculated after an AudioSendStream has been created. This means that before an O/A negotiation, audio levels are unavailable. According to spec, if we have an audio source, we should have audio levels. An immediate solution to this would have been to calculate the audio level at pc/rtp_sender.cc. The problem is that the LocalAudioSinkAdapter::OnData() code path, while exercised in chromium, is not exercised in native code. The issue of calculating audio levels on a per-source bases rather than on a per-send stream basis is left to https://crbug.com/webrtc/10771, an existing "media-source" bug. This CL can be verified manually in Chrome at: https://codepen.io/anon/pen/vqRGyq Bug: chromium:736403, webrtc:10771 Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-03 17:11:10 +02:00
testonly = true
sources = [
"audio_frame_manipulator_unittest.cc",
"audio_mixer_impl_unittest.cc",
"frame_combiner_unittest.cc",
]
deps = [
":audio_frame_manipulator",
":audio_mixer_impl",
":audio_mixer_test_utils",
"../../api:array_view",
BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-16 13:43:49 +01:00
"../../api/audio:audio_mixer_api",
Send data from mixer to APM limiter more often. Before this change, the APM limiter used in FrameCombiner (a sub-component of AudioMixer) only gets to process the data when the number of non-muted streams is >1. If this number varies between <=1 and >1, the limiter's view of the data will have gaps during the periods with <= 1 active stream. This leads to discontinuities in the applied gain. These discontinuities cause clicks in the output audio. This change activates APM limiter processing based on the number of audio streams, independently of their mutedness status. BUG=chromium:695993 Review-Url: https://codereview.webrtc.org/2776113002 Cr-Commit-Position: refs/heads/master@{#17442}
2017-03-29 04:25:16 -07:00
"../../audio/utility:audio_frame_operations",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Avoid using GlobalTaskQueueFactory for TaskQueueForTest To remove global task factory, rtc::TaskQueue need to loose it's convenient constructor TaskQueueForTest can be used instead in tests and keep the convenient constructor. Also cleanup the TaskQueueForTest a bit: move the class to webrtc namespace add default constructor disallow copy using language construct instead of macro cleanup build dependencies rename build target (to match move out of the rtc namespace) Bug: webrtc:10284 Change-Id: I17fddf3f8d4f363df7d495c28a5b0a28abda1ba7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127571 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27193}
2019-03-19 18:08:37 +01:00
"../../rtc_base:task_queue_for_test",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:test_support",
]
}
New test binary for the AudioMixer. Allows mixing up to 4 input streams. Useful for profiling and manual tests. Allows testing different combinations of input/output rates and number of channels. Reads and writes WAV files. Can also configure whether to use the Limiter component of the AudioMixer. Bug: webrtc:8925 Change-Id: Iaf4fee5284980f6ed01f4bb721e49bb1af8dd392 Reviewed-on: https://webrtc-review.googlesource.com/56842 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22209}
2018-02-27 14:09:47 +01:00
rtc_executable("audio_mixer_test") {
testonly = true
sources = [
"audio_mixer_test.cc",
]
deps = [
":audio_mixer_impl",
"../../api/audio:audio_mixer_api",
"../../common_audio",
Migrate WebRTC test infra to ABSL_FLAG. This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag will be removed soon after this one lands. Bug: webrtc:10616 Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28606}
2019-07-18 13:44:12 +02:00
"../../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
New test binary for the AudioMixer. Allows mixing up to 4 input streams. Useful for profiling and manual tests. Allows testing different combinations of input/output rates and number of channels. Reads and writes WAV files. Can also configure whether to use the Limiter component of the AudioMixer. Bug: webrtc:8925 Change-Id: Iaf4fee5284980f6ed01f4bb721e49bb1af8dd392 Reviewed-on: https://webrtc-review.googlesource.com/56842 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22209}
2018-02-27 14:09:47 +01:00
]
}
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
}
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