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webrtc_m130/modules/audio_mixer/BUILD.gn

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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") {
public_deps = [
":audio_frame_manipulator",
":audio_mixer_impl",
]
}
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
rtc_static_library("audio_mixer_impl") {
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
]
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
public_deps = [
"../../api:audio_mixer_api",
]
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",
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",
Added new mixer to |check_targets| in .gn and fixed include/depend errors. Also fixed one small chromium-style error in the new mixer. NOTRY=True Committed: https://crrev.com/d700bef583d29ba2834ae57b3af7e8d3b8306cb9 Review-Url: https://codereview.webrtc.org/2234293002 Cr-Original-Commit-Position: refs/heads/master@{#13752} Cr-Commit-Position: refs/heads/master@{#13772}
2016-08-16 02:15:49 -07:00
"../..:webrtc_common",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../../api:array_view",
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",
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",
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",
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
]
}
rtc_static_library("audio_frame_manipulator") {
visibility = [
":*",
"../../modules:*",
]
sources = [
"audio_frame_manipulator.cc",
"audio_frame_manipulator.h",
]
deps = [
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",
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",
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) {
rtc_source_set("audio_mixer_unittests") {
testonly = true
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
sources = [
"audio_frame_manipulator_unittest.cc",
"audio_mixer_impl_unittest.cc",
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_unittest.cc",
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
]
deps = [
":audio_frame_manipulator",
":audio_mixer_impl",
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",
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",
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
"../../api: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",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_task_queue",
GN: 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",
"//testing/gmock",
]
}
}
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