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webrtc_m130/audio/BUILD.gn

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Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
# Copyright (c) 2015 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")
Adding placeholder for low bandwidth audio test This will allow the trybots to be updated to start running this new test executable, so that they can be used when landing this CL which will replace the dummy test with real code: https://codereview.webrtc.org/2694203002 Most likely, the trybots will just run the test binary while the perf bots will run a Python wrapper script that takes care of the post-processing to calculate audio quality using PESQ. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2717683002 Cr-Commit-Position: refs/heads/master@{#17063}
2017-03-06 04:01:16 -08:00
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
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") {
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
sources = [
Move remaining traces of VoiceEngine - Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 11:18:31 +01:00
"audio_level.cc",
"audio_level.h",
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
"audio_receive_stream.cc",
"audio_receive_stream.h",
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1397123003 Cr-Commit-Position: refs/heads/master@{#10307}
2015-10-16 14:35:07 -07:00
"audio_send_stream.cc",
"audio_send_stream.h",
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1403363003 Cr-Commit-Position: refs/heads/master@{#10548}
2015-11-06 15:34:49 -08:00
"audio_state.cc",
"audio_state.h",
Remove voe::TransmitMixer TransmitMixer's functionality is moved into the AudioTransportProxy owned by AudioState. This removes the need for an AudioTransport implementation in VoEBaseImpl, which means that the proxy is no longer a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl. In the short term, AudioState needs to know which AudioDeviceModule is used, so it is added in AudioState::Config. AudioTransportImpl needs to know which AudioSendStream:s are currently enabled to send, so AudioState maintains a map of them, which is reduced into a simple vector for AudioTransportImpl. To encode and transmit audio, AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame) is introduced, which is used in both the Chromium and standalone use cases. This removes the need for two different instances of voe::Channel::ProcessAndEncodeAudio(), so there is now only one, taking an AudioFrame as argument. Callers need to allocate their own AudioFrame:s, which is wasteful but not a regression since this was already happening in the voe::Channel functions. Most of the logic changed resides in AudioTransportImpl::RecordedDataIsAvailable(), where two strange things were found: 1. The clock drift parameter was ineffective since apm->echo_cancellation()->enable_drift_compensation(false) is called during initialization. 2. The output parameter 'new_mic_volume' was never set - instead it was returned as a result, causing the ADM to never update the analog mic gain (https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100). Besides this, tests are updated, and some dead code is removed which was found in the process. Bug: webrtc:4690, webrtc:8591 Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2 Reviewed-on: https://webrtc-review.googlesource.com/26681 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
"audio_transport_impl.cc",
"audio_transport_impl.h",
Split voe::Channel into ChannelSend and ChannelReceive Bug: webrtc:9801 Change-Id: Ia15af1e53c8d384ad6e5fbddcb25311fce4befae Reviewed-on: https://webrtc-review.googlesource.com/c/103640 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24992}
2018-10-04 14:28:39 +02:00
"channel_receive.cc",
"channel_receive.h",
"channel_send.cc",
"channel_send.h",
Re-Land: Implement AudioReceiveStream::GetStats(). R=tommi@webrtc.org BUG=webrtc:4690 Committed: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0 Review URL: https://codereview.webrtc.org/1390753002 . Cr-Commit-Position: refs/heads/master@{#10369}
2015-10-22 10:49:27 +02:00
"conversion.h",
Revert "Inlines NullAudioPoller functionality into AudioState class." This reverts commit 0e96535be97916d8fcaa9873ffab3c636539f9d8. Reason for revert: Downstream test failure Original change's description: > Inlines NullAudioPoller functionality into AudioState class. > > As part of this, we also use TaskQueue and RepeatedTask rather > than rtc::Thread + rtc::MessageHandler. With the ultimate goal of > deprecating rtc::Thread. > > Bug: webrtc:9883 > Change-Id: I2fb851ac31ee2431435d51de78ff446572512201 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167528 > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30430} TBR=saza@webrtc.org,srte@webrtc.org Change-Id: I4c77259f7b6477fc1cb79350f2d47817f106770d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168046 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30431}
2020-01-30 18:13:54 +00:00
"null_audio_poller.cc",
"null_audio_poller.h",
Move remaining traces of VoiceEngine - Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 11:18:31 +01:00
"remix_resample.cc",
"remix_resample.h",
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
]
deps = [
Move remaining traces of VoiceEngine - Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 11:18:31 +01:00
"../api:array_view",
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
"../api:call_api",
Insert audio frame transformer between encoder and packetizer. The frame transformer is passed from RTPSenderInterface through the library to be eventually set in ChannelSend, where the frame transformation will occur in the follow-up CL. Insertable Streams Web API explainer: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I01b2adc3c96b948d182d5401a9a4fe14cf5960a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171870 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30946}
2020-03-31 11:29:56 +02:00
"../api:frame_transformer_interface",
Move rtc::FunctionView to the public API Bug: webrtc:10138 Change-Id: Icc25a2a277a9608701aaddd546882366739991ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27227}
2019-03-21 14:37:36 +01:00
"../api:function_view",
Reland "Move api/rtp_headers.h to its own build target." This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e Original change's description: > Move api/rtp_headers.h to its own build target. > > Reduces dependencies on the libjingle_peerconnection_api target from > lower-level code. > > Bug: None > Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27213} Bug: None Tbr: kwiberg@webrtc.org Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-20 08:40:23 +01:00
"../api:rtp_headers",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
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",
Move remaining traces of VoiceEngine - Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 11:18:31 +01:00
"../api:transport_api",
Separate build targets for aec3 and aec3_unittests Bug: webrtc:8844 Change-Id: Id6a98eae19aaedc87c3f402a004f58f0290d5c28 Reviewed-on: https://webrtc-review.googlesource.com/56580 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22173}
2018-02-23 13:18:29 +01:00
"../api/audio:aec3_factory",
Remove dependencies on modules:module_api from AudioProcessing. - Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
2018-04-12 22:44:09 +02:00
"../api/audio:audio_frame_api",
BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-16 13:43:49 +01:00
"../api/audio:audio_mixer_api",
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2799033006 Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 06:46:30 -07:00
"../api/audio_codecs:audio_codecs_api",
Introduce api/crypto/BUILD.gn. No-Try: True Bug: webrtc:8733 Change-Id: I8679735be1e5069e371a9f1115a54e897e09964b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152622 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29186}
2019-09-13 14:42:15 +02:00
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
Enable injection of a custom NetEqFactory into PeerConnectionFactory. Injecting both a custom NetEqFactory and an AudioDecoderFactory is not supported, in that case the AudioDecoderFactory should be wrapped inside the NetEqFactory. Bug: webrtc:11005 Change-Id: I4e311eb1bfa03c91bca587d70540e81829f881c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158720 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29673}
2019-11-01 11:47:51 +01:00
"../api/neteq:neteq_api",
Delete deprecated rtc_event_log header Bug: webrtc:10206 Change-Id: I9ed3148843c647372993729b87c0e74741ab540b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 12:24:53 +02:00
"../api/rtc_event_log",
Don't inject worker queue into send streams. This prepares for making AudioSendStream use its own task queue. In the future more of the functionality that depends on running on the task queue is planned to be moved directly into RtpTransportControllerSend. They should instead get it from the transport controller. This affects the media transport tests which previously assumed that the transport controller could be missing. However, this is not something that is used in production, so this is an improvement of the tests as they will behave more like production code. Bug: webrtc:9883 Change-Id: Ie32f4c2f6433ec37ac16a08d531ceb690ea9c0b5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126000 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27010}
2019-03-07 09:17:19 +01:00
"../api/task_queue",
Split out RtpSource from libjingle_peerconnection_api And moved declaration into a new api directory, as api/transport/rtp/rtp_source.h. Bug: webrtc:8733 Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29039}
2019-09-02 15:16:49 +02:00
"../api/transport/rtp:rtp_source",
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api Bug: webrtc:11251 Change-Id: Id3b6ff1814931d8250c4aaac59e494521fbe93ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164560 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tim Na <natim@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30238}
2020-01-10 10:33:05 -08:00
"../call:audio_sender_interface",
Fix Gn untracked headers in webrtc/call. This CL is the same CL we had at https://codereview.webrtc.org/3014543002/. Since we cannot land it with Rietveld anymore let's move the discussion to Gerrit. BUG=webrtc:7641 CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Change-Id: I5662bec318544b07f476c12ecada997d726e7361 Reviewed-on: https://webrtc-review.googlesource.com/7981 Reviewed-by: Henrik Kjellander <kjellander@google.com> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20224}
2017-10-10 14:38:11 +02:00
"../call:bitrate_allocator",
Moved call.h and most of api/call/* into call/ BUG=webrtc:6716 Review-Url: https://codereview.webrtc.org/2550273003 Cr-Commit-Position: refs/heads/master@{#15460}
2016-12-07 04:52:58 -08:00
"../call:call_interfaces",
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. BUG=webrtc:7135 TBR=sprang@webrtc.org Review-Url: https://codereview.webrtc.org/2913143003 Cr-Commit-Position: refs/heads/master@{#18371}
2017-06-01 04:02:35 -07:00
"../call:rtp_interfaces",
Replace AudioConferenceMixer with AudioMixer. This CL re-routes audio through AudioMixer instead of AudioConferenceMixer. This is done without any modifications to VoiceEngine. Previously, output audio was polled by an AudioDevice through an AudioTransport pointer, which was an instance of VoEBaseImpl. VoiceEngineImpl sent the request for data on to OutputMixer and further to AudioConferenceMixer. This CL changes the audio flow to an AudioDevice. We reconfigure the AudioDevice to have another AudioTransport pointer, which points to an AudioTransportProxy. The AudioTransportProxy is responsible for feeding mixed data to the AudioProcessing component for echo cancellation, and to resample the audio data after AudioProcessing and before it is sent to the AudioDevice. The set up of the audio path was previously done during VoiceEngine initialization. Now it is changed in the AudioState constructor. This list shows where audio-path-related VoiceEngine functionality has been moved: OutputMixer --> AudioTransportProxy VoiceEngineImpl --> AudioState, AudioTransportProxy SharedData --> AudioState Channel --> AudioReceiveStream, ChannelProxy, Channel AudioState owns the new mixer and connects it to AudioTransport and AudioDevice on initialization. The audio input source is AudioReceiveStream, which registers itself with the mixer (which it gets from AudioState) on Start and Stop. # Since the AudioTransport interface contains non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2436033002 Cr-Commit-Position: refs/heads/master@{#15193}
2016-11-22 06:42:53 -08:00
"../common_audio",
Stop using public_deps in common_audio. Bug: webrtc:8603 Change-Id: I315311977f2a75476a7028b8d3eaf3c98caf4178 Reviewed-on: https://webrtc-review.googlesource.com/47920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21902}
2018-02-05 15:50:41 +01:00
"../common_audio:common_audio_c",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../logging:rtc_event_audio",
Move event logging of config into AudioSendStream. It was previously logged in Call, but streams are not always created with the full configuration, which caused header extensions to be missing from the log. Bug: webrtc:9885 Change-Id: I86c0424004c4629ebab0f6b155b83fb90e15b131 Reviewed-on: https://webrtc-review.googlesource.com/c/108601 Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25483}
2018-10-30 16:11:02 +01:00
"../logging:rtc_stream_config",
Move remaining traces of VoiceEngine - Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 11:18:31 +01:00
"../modules/audio_coding",
In ChannelReceive, use AcmReceiver directly, not AudioCodingModule Bug: webrtc:9801 Change-Id: I02d76bc89c363247c8dc782db316a9f87a2b93ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/111504 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28766}
2019-07-30 15:15:59 +02:00
"../modules/audio_coding:audio_coding_module_typedefs",
Hide the AudioEncoderCng class behind a create function And put codecs/cng/webrtc_cng.h in a non-public build target while we're at it. Bug: webrtc:8396 Change-Id: I9f51dffadfb645cd1454617fad30e09d639ff53c Reviewed-on: https://webrtc-review.googlesource.com/c/108782 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25486}
2018-11-01 11:13:44 +01:00
"../modules/audio_coding:audio_encoder_cng",
Move remaining traces of VoiceEngine - Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 11:18:31 +01:00
"../modules/audio_coding:audio_network_adaptor_config",
Added an empty AudioTransportProxy to AudioState. All audio in calls is now routed through AudioTransportProxy. The AudioTransport implemented by VoEBaseImpl is disconnected from AudioDevice and replaced by an empty proxy layer that forwards calls to the old Transport. This is a refactoring CL in preparation for landing https://codereview.webrtc.org/2436033002/, which will connect the new AudioMixer. In the planned configuration, the currently empty AudioTransportProxy will query the new mixer for audio instead of polling data from the old Transport. Mixed audio will be passed to an AudioProcessing interface. AudioTransportProxy is initialized with an AudioProcessing*, which is currently unused. No presubmit since we implement an interface with non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2454373002 Cr-Commit-Position: refs/heads/master@{#15133}
2016-11-17 06:28:59 -08:00
"../modules/audio_device",
"../modules/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
"../modules/audio_processing:api",
APM: Localize/abstract the usage of AudioFrame This CL moves the implementation of of the AudioFrame support from the implementation of AudioProcessing to proxy methods that map the call to the integer stream interfaces (added in another CL). The CL also changes the WebRTC code using the AudioFrame interfaces to instead use the proxy methods. This CL will be followed by one more CL that removes the usage of the AudioFrame class from the rest of APM (apart from the AudioProcessing API). Bug: webrtc:5298 Change-Id: Iecb72e9fa896ebea3ac30e558489c1bac88f5891 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170110 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30812}
2020-03-17 13:23:58 +01:00
"../modules/audio_processing:audio_frame_proxies",
Split RMS level measurement utility from APM This moves rms_level.* into a separate build target. Bug: webrtc:11226 Change-Id: I94ceacd1ec65dda48f5d19b22ba2625d13543e08 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170323 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30856}
2020-03-20 15:55:29 +01:00
"../modules/audio_processing:rms_level",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../modules/pacing",
"../modules/remote_bitrate_estimator",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../modules/rtp_rtcp",
Move remaining traces of VoiceEngine - Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 11:18:31 +01:00
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/utility",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../rtc_base",
This CL removes all usages of our custom ostream << overloads. This prepares us for removing them altogether. Bug: webrtc:8982 Change-Id: I66002cc8d4bf0e07925766d568d2498422f0f38e Reviewed-on: https://webrtc-review.googlesource.com/64142 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Henrik Grunell <henrikg@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22707}
2018-04-03 13:40:05 +02:00
"../rtc_base:audio_format_to_string",
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",
Move remaining traces of VoiceEngine - Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 11:18:31 +01:00
"../rtc_base:rate_limiter",
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",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../rtc_base:safe_minmax",
Removing AudioAllocationSettings moving functionality to AudioSendStream. This is a no-op change that just removes the AudioAllocationSettings helper class that was previously introduced since the field trials in it were used in several places. Those other usages has now been removed and AudioSendStream is now the only user. By moving the trials directly to AudioSendStream we reduce the reader overhead when trying to follow what a particular field trial does. The "WebRTC-Audio-ForceNoTWCC" trial was removed as it is always set together with "WebRTC-Audio-ABWENoTWCC". Bug: webrtc:9883 Change-Id: Ib63589255bfe7adb155ea41279bdcd153f1536c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155366 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29371}
2019-10-03 10:03:55 +02:00
"../rtc_base/experiments:field_trial_parser",
Insert audio frame transformer between encoder and packetizer. The frame transformer is passed from RTPSenderInterface through the library to be eventually set in ChannelSend, where the frame transformation will occur in the follow-up CL. Insertable Streams Web API explainer: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I01b2adc3c96b948d182d5401a9a4fe14cf5960a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171870 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30946}
2020-03-31 11:29:56 +02:00
"../rtc_base/synchronization:sequence_checker",
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
"../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:field_trial",
"../system_wrappers:metrics",
Remove voe::TransmitMixer TransmitMixer's functionality is moved into the AudioTransportProxy owned by AudioState. This removes the need for an AudioTransport implementation in VoEBaseImpl, which means that the proxy is no longer a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl. In the short term, AudioState needs to know which AudioDeviceModule is used, so it is added in AudioState::Config. AudioTransportImpl needs to know which AudioSendStream:s are currently enabled to send, so AudioState maintains a map of them, which is reduced into a simple vector for AudioTransportImpl. To encode and transmit audio, AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame) is introduced, which is used in both the Chromium and standalone use cases. This removes the need for two different instances of voe::Channel::ProcessAndEncodeAudio(), so there is now only one, taking an AudioFrame as argument. Callers need to allocate their own AudioFrame:s, which is wasteful but not a regression since this was already happening in the voe::Channel functions. Most of the logic changed resides in AudioTransportImpl::RecordedDataIsAvailable(), where two strange things were found: 1. The clock drift parameter was ineffective since apm->echo_cancellation()->enable_drift_compensation(false) is called during initialization. 2. The output parameter 'new_mic_volume' was never set - instead it was returned as a result, causing the ADM to never update the analog mic gain (https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100). Besides this, tests are updated, and some dead code is removed which was found in the process. Bug: webrtc:4690, webrtc:8591 Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2 Reviewed-on: https://webrtc-review.googlesource.com/26681 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
"utility:audio_frame_operations",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace rtc::Optional with absl::optional in audio, call and video This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameters 'audio call video': #!/bin/bash find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: I02c5db956846a88a268a300ba086703a02d62e36 Reviewed-on: https://webrtc-review.googlesource.com/83722 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23628}
2018-06-15 12:28:07 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
2015-09-25 13:58:30 +02:00
]
}
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
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_end_to_end_test") {
- Removes voe_conference_test. - Adds a new AudioStatsTest, with better coverage of the same features, based on call_test. - Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses. BUG=webrtc:4690 R=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/3008273002 . Cr-Commit-Position: refs/heads/master@{#19833}
2017-09-14 14:46:47 +02:00
testonly = true
sources = [
"test/audio_end_to_end_test.cc",
"test/audio_end_to_end_test.h",
]
deps = [
":audio",
Introduce SimulatedNetworkReceiverInterface. Introduce SimulatedNetworkReceiverInterface and switch DirectTransport on this interface. Also switch part of related users on DefaultNetworkSimulationConfig. This two changes united into single CL to prevent work duplication. Most changes were done because of stop including fake_network_pipe.h into direct_transport.h, so splitting this into 2 CLs will require first fix all imports of fake_network_pipe.h and then replace them on new API imports again. Bug: webrtc:9630 Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6 Reviewed-on: https://webrtc-review.googlesource.com/94762 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24336}
2018-08-17 14:26:54 +02:00
"../api:simulated_network_api",
Propagate task queue to create test::DirectTransport by TaskQueueBase interface actual task queue implementation for these tests is intentionally unchanged for now. while at it, change return type of created transports to unique_ptr to note passing ownership. Bug: webrtc:10933 Change-Id: I324597b503e647c471f43511340eb9c07ba03ee8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154743 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29335}
2019-09-30 04:16:28 +02:00
"../api/task_queue",
Mark DirectTransport subclasses ctors as deprecated and switch from them Bug: webrtc:9630 Change-Id: I6e7bf898fd95ef76758458e759d9f9aa381f89e1 Reviewed-on: https://webrtc-review.googlesource.com/94843 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24345}
2018-08-20 13:30:39 +02:00
"../call:fake_network",
"../call:simulated_network",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../system_wrappers",
- Removes voe_conference_test. - Adds a new AudioStatsTest, with better coverage of the same features, based on call_test. - Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses. BUG=webrtc:4690 R=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/3008273002 . Cr-Commit-Position: refs/heads/master@{#19833}
2017-09-14 14:46:47 +02:00
"../test:test_common",
"../test:test_support",
]
}
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_tests") {
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
testonly = true
Refactor webrtc/{api,audio} and modules/audio_coding for GN check This moves some GN check configurations out of .gn to individual targets. The now checked targets are: "//webrtc/api/*", "//webrtc/audio/*", "//webrtc/modules/audio_coding/*", Many targets were fixed by adding dependencies, but the ones that requires more refactorings are left with the check_includes attribute set to false instead. Make //webrtc/test:test_support a public dep of //webrtc/test:test_main to avoid having to add that to all users of it. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2556943003 Cr-Commit-Position: refs/heads/master@{#15461}
2016-12-07 08:23:27 -08:00
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
sources = [
"audio_receive_stream_unittest.cc",
Remove voe_auto_test and add new tests to cover the missing cases. BUG=webrtc:4690 Review-Url: https://codereview.webrtc.org/3007383002 Cr-Commit-Position: refs/heads/master@{#19865}
2017-09-15 09:56:08 -07:00
"audio_send_stream_tests.cc",
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
"audio_send_stream_unittest.cc",
"audio_state_unittest.cc",
Move remaining traces of VoiceEngine - Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 11:18:31 +01:00
"mock_voe_channel_proxy.h",
"remix_resample_unittest.cc",
Remove memcheck. Since the linux_memcheck trybot is no more, this CL removes all the code needed to make it work. Bug: webrtc:7737, webrtc:8356, webrtc:9570 Change-Id: I09a9467b8bf895146a3384c2c915b54662721af6 Reviewed-on: https://webrtc-review.googlesource.com/90863 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24201}
2018-08-07 08:53:41 +02:00
"test/audio_stats_test.cc",
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
]
deps = [
":audio",
- Removes voe_conference_test. - Adds a new AudioStatsTest, with better coverage of the same features, based on call_test. - Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses. BUG=webrtc:4690 R=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/3008273002 . Cr-Commit-Position: refs/heads/master@{#19833}
2017-09-14 14:46:47 +02:00
":audio_end_to_end_test",
Introduce MediaTransportConfig Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files. TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change. Bug: webrtc:9719 Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28016}
2019-05-21 11:12:57 -07:00
"../api:libjingle_peerconnection_api",
Replace AudioConferenceMixer with AudioMixer. This CL re-routes audio through AudioMixer instead of AudioConferenceMixer. This is done without any modifications to VoiceEngine. Previously, output audio was polled by an AudioDevice through an AudioTransport pointer, which was an instance of VoEBaseImpl. VoiceEngineImpl sent the request for data on to OutputMixer and further to AudioConferenceMixer. This CL changes the audio flow to an AudioDevice. We reconfigure the AudioDevice to have another AudioTransport pointer, which points to an AudioTransportProxy. The AudioTransportProxy is responsible for feeding mixed data to the AudioProcessing component for echo cancellation, and to resample the audio data after AudioProcessing and before it is sent to the AudioDevice. The set up of the audio path was previously done during VoiceEngine initialization. Now it is changed in the AudioState constructor. This list shows where audio-path-related VoiceEngine functionality has been moved: OutputMixer --> AudioTransportProxy VoiceEngineImpl --> AudioState, AudioTransportProxy SharedData --> AudioState Channel --> AudioReceiveStream, ChannelProxy, Channel AudioState owns the new mixer and connects it to AudioTransport and AudioDevice on initialization. The audio input source is AudioReceiveStream, which registers itself with the mixer (which it gets from AudioState) on Start and Stop. # Since the AudioTransport interface contains non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2436033002 Cr-Commit-Position: refs/heads/master@{#15193}
2016-11-22 06:42:53 -08:00
"../api:mock_audio_mixer",
Fixes use after free error when setting a new FrameEncryptor on ChannelSend. This change corrects a potential race condition when updating a FrameEncryptor for the audio send channel. If a FrameEncryptor is set on an active audio stream it is possible for the current FrameEncryptor attached to the audio channel to be deallocated due to the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the channel. To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor> to only allow deallocation when it is actually set on the encoder queue. ChannelSend is unique in this respect as the Audio Receiver a long with the Video Sender and Video Receiver streams all recreate themselves when they have a configuration change. ChannelSend instead reconfigures itself using the existing channel object. Added Seth as TBR as this only introduces mocks. TBR=shampson@webrtc.org Bug: webrtc:9907 Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4 Reviewed-on: https://webrtc-review.googlesource.com/c/107664 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25374}
2018-10-25 09:52:57 -07:00
"../api:mock_frame_decryptor",
"../api:mock_frame_encryptor",
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",
Use MediaTransportInterface, for audio streams. Bug: webrtc:9719 Change-Id: I6d3db66b781173b207de51d84193fbd34a7f3239 Reviewed-on: https://webrtc-review.googlesource.com/c/104642 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25385}
2018-10-26 12:57:07 +02:00
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs/opus:audio_decoder_opus",
"../api/audio_codecs/opus:audio_encoder_opus",
Delete deprecated rtc_event_log header Bug: webrtc:10206 Change-Id: I9ed3148843c647372993729b87c0e74741ab540b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 12:24:53 +02:00
"../api/rtc_event_log",
Avoid using global task queue factory in audio/ unittests in particular replace rtc::TaskQueue with TaskQueueForTest class since latter uses DefaultTaskQueueFactory() directly instead of through GlobalTaskQueueFactory Bug: webrtc:10284 Change-Id: I1a52c5942626e3e2256b3d78975d2740e9facb1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128880 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27245}
2019-03-22 12:59:48 +01:00
"../api/task_queue:default_task_queue_factory",
Replacing rtc::TimeDelta with webrtc::TimeDelta. This removes the redundant type and replaces all usages. A slight change in behavior is that we no longer get nanosecond resolution. This should not matter since no current code requires nanosecond resolution. Bug: webrtc:9155 Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2 Reviewed-on: https://webrtc-review.googlesource.com/71163 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23174}
2018-05-08 14:52:22 +02:00
"../api/units:time_delta",
Use MediaTransportInterface, for audio streams. Bug: webrtc:9719 Change-Id: I6d3db66b781173b207de51d84193fbd34a7f3239 Reviewed-on: https://webrtc-review.googlesource.com/c/104642 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25385}
2018-10-26 12:57:07 +02:00
"../call:mock_bitrate_allocator",
Remove voe::TransmitMixer TransmitMixer's functionality is moved into the AudioTransportProxy owned by AudioState. This removes the need for an AudioTransport implementation in VoEBaseImpl, which means that the proxy is no longer a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl. In the short term, AudioState needs to know which AudioDeviceModule is used, so it is added in AudioState::Config. AudioTransportImpl needs to know which AudioSendStream:s are currently enabled to send, so AudioState maintains a map of them, which is reduced into a simple vector for AudioTransportImpl. To encode and transmit audio, AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame) is introduced, which is used in both the Chromium and standalone use cases. This removes the need for two different instances of voe::Channel::ProcessAndEncodeAudio(), so there is now only one, taking an AudioFrame as argument. Callers need to allocate their own AudioFrame:s, which is wasteful but not a regression since this was already happening in the voe::Channel functions. Most of the logic changed resides in AudioTransportImpl::RecordedDataIsAvailable(), where two strange things were found: 1. The clock drift parameter was ineffective since apm->echo_cancellation()->enable_drift_compensation(false) is called during initialization. 2. The output parameter 'new_mic_volume' was never set - instead it was returned as a result, causing the ADM to never update the analog mic gain (https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100). Besides this, tests are updated, and some dead code is removed which was found in the process. Bug: webrtc:4690, webrtc:8591 Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2 Reviewed-on: https://webrtc-review.googlesource.com/26681 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
"../call:mock_call_interfaces",
Using mock transport controller in audio unit tests. Using a mock of rtp transport controller send in audio send stream unit tests. This reduces the dependencies and makes the tests more focused on testing the functionality of audio send stream itself. Bug: webrtc:8415 Change-Id: Ia8d9cf47d93decc74b10ca75a6771f39df658dc2 Reviewed-on: https://webrtc-review.googlesource.com/56600 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22161}
2018-02-22 14:49:02 +01:00
"../call:mock_rtp_interfaces",
Fix deps of audio:audio_tests. Bug: webrtc:6828 Change-Id: Iae9020fda37fe40221d9a9def38c3afcc387d359 Reviewed-on: https://webrtc-review.googlesource.com/22683 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20666}
2017-11-13 17:04:05 +01:00
"../call:rtp_interfaces",
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. And implementation class RtpStreamReceiverController. It's responsible for demuxing, and acts as factory for RtpStreamReceiverInterface. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2886993005 Cr-Commit-Position: refs/heads/master@{#18696}
2017-06-21 01:05:22 -07:00
"../call:rtp_receiver",
Don't inject worker queue into send streams. This prepares for making AudioSendStream use its own task queue. In the future more of the functionality that depends on running on the task queue is planned to be moved directly into RtpTransportControllerSend. They should instead get it from the transport controller. This affects the media transport tests which previously assumed that the transport controller could be missing. However, this is not something that is used in production, so this is an improvement of the tests as they will behave more like production code. Bug: webrtc:9883 Change-Id: Ie32f4c2f6433ec37ac16a08d531ceb690ea9c0b5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126000 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27010}
2019-03-07 09:17:19 +01:00
"../call:rtp_sender",
Move remaining traces of VoiceEngine - Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 11:18:31 +01:00
"../common_audio",
Reland: Add mock_rtc_event_log.h. Bug: webrtc:7642 Change-Id: I3f97a8b603e34819e1563b335c1eeb51f89b11ac Reviewed-on: https://webrtc-review.googlesource.com/37081 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21480}
2018-01-03 09:08:20 +01:00
"../logging:mocks",
Avoid using global task queue factory in audio/ unittests in particular replace rtc::TaskQueue with TaskQueueForTest class since latter uses DefaultTaskQueueFactory() directly instead of through GlobalTaskQueueFactory Bug: webrtc:10284 Change-Id: I1a52c5942626e3e2256b3d78975d2740e9facb1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128880 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27245}
2019-03-22 12:59:48 +01:00
"../modules/audio_device:audio_device_impl", # For TestAudioDeviceModule
Added an empty AudioTransportProxy to AudioState. All audio in calls is now routed through AudioTransportProxy. The AudioTransport implemented by VoEBaseImpl is disconnected from AudioDevice and replaced by an empty proxy layer that forwards calls to the old Transport. This is a refactoring CL in preparation for landing https://codereview.webrtc.org/2436033002/, which will connect the new AudioMixer. In the planned configuration, the currently empty AudioTransportProxy will query the new mixer for audio instead of polling data from the old Transport. Mixed audio will be passed to an AudioProcessing interface. AudioTransportProxy is initialized with an AudioProcessing*, which is currently unused. No presubmit since we implement an interface with non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2454373002 Cr-Commit-Position: refs/heads/master@{#15133}
2016-11-17 06:28:59 -08:00
"../modules/audio_device:mock_audio_device",
Replace AudioConferenceMixer with AudioMixer. This CL re-routes audio through AudioMixer instead of AudioConferenceMixer. This is done without any modifications to VoiceEngine. Previously, output audio was polled by an AudioDevice through an AudioTransport pointer, which was an instance of VoEBaseImpl. VoiceEngineImpl sent the request for data on to OutputMixer and further to AudioConferenceMixer. This CL changes the audio flow to an AudioDevice. We reconfigure the AudioDevice to have another AudioTransport pointer, which points to an AudioTransportProxy. The AudioTransportProxy is responsible for feeding mixed data to the AudioProcessing component for echo cancellation, and to resample the audio data after AudioProcessing and before it is sent to the AudioDevice. The set up of the audio path was previously done during VoiceEngine initialization. Now it is changed in the AudioState constructor. This list shows where audio-path-related VoiceEngine functionality has been moved: OutputMixer --> AudioTransportProxy VoiceEngineImpl --> AudioState, AudioTransportProxy SharedData --> AudioState Channel --> AudioReceiveStream, ChannelProxy, Channel AudioState owns the new mixer and connects it to AudioTransport and AudioDevice on initialization. The audio input source is AudioReceiveStream, which registers itself with the mixer (which it gets from AudioState) on Start and Stop. # Since the AudioTransport interface contains non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2436033002 Cr-Commit-Position: refs/heads/master@{#15193}
2016-11-22 06:42:53 -08:00
"../modules/audio_mixer:audio_mixer_impl",
[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
"../modules/audio_mixer:audio_mixer_test_utils",
Use the new AudioProcessing statistics everywhere. The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 17:29:59 +01:00
"../modules/audio_processing:audio_processing_statistics",
Add missing files to audio_processing. Bug: webrtc:7621 Change-Id: I2cab764232fc4e084ed8a489f4cf3a3ac562c894 Reviewed-on: https://webrtc-review.googlesource.com/34658 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21461}
2017-12-19 16:44:45 +01:00
"../modules/audio_processing:mocks",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../modules/pacing",
Fix deps of audio:audio_tests. Bug: webrtc:6828 Change-Id: Iae9020fda37fe40221d9a9def38c3afcc387d359 Reviewed-on: https://webrtc-review.googlesource.com/22683 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20666}
2017-11-13 17:04:05 +01:00
"../modules/rtp_rtcp:mock_rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
Use MediaTransportInterface, for audio streams. Bug: webrtc:9719 Change-Id: I6d3db66b781173b207de51d84193fbd34a7f3239 Reviewed-on: https://webrtc-review.googlesource.com/c/104642 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25385}
2018-10-26 12:57:07 +02:00
"../modules/utility",
Move remaining traces of VoiceEngine - Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 11:18:31 +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 global task queue factory in audio/ unittests in particular replace rtc::TaskQueue with TaskQueueForTest class since latter uses DefaultTaskQueueFactory() directly instead of through GlobalTaskQueueFactory Bug: webrtc:10284 Change-Id: I1a52c5942626e3e2256b3d78975d2740e9facb1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128880 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27245}
2019-03-22 12:59:48 +01:00
"../rtc_base:rtc_base_tests_utils",
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:safe_compare",
Avoid using global task queue factory in audio/ unittests in particular replace rtc::TaskQueue with TaskQueueForTest class since latter uses DefaultTaskQueueFactory() directly instead of through GlobalTaskQueueFactory Bug: webrtc:10284 Change-Id: I1a52c5942626e3e2256b3d78975d2740e9facb1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128880 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27245}
2019-03-22 12:59:48 +01:00
"../rtc_base:task_queue_for_test",
Add some missing includes and dependencies. In particular, time_utils.h is currently pulled in via rtc_event.h This CL is in preparation of moving parts of the RTC event log to api/. Bug: webrtc:10206 Change-Id: Idd35aa9404afded4d29b1296344996c45b8c2e91 Reviewed-on: https://webrtc-review.googlesource.com/c/117921 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26326}
2019-01-16 17:45:05 +01:00
"../rtc_base:timeutils",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../system_wrappers",
Fix deps of audio:audio_tests. Bug: webrtc:6828 Change-Id: Iae9020fda37fe40221d9a9def38c3afcc387d359 Reviewed-on: https://webrtc-review.googlesource.com/22683 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20666}
2017-11-13 17:04:05 +01:00
"../test:audio_codec_mocks",
Avoid using global task queue factory in audio/ unittests in particular replace rtc::TaskQueue with TaskQueueForTest class since latter uses DefaultTaskQueueFactory() directly instead of through GlobalTaskQueueFactory Bug: webrtc:10284 Change-Id: I1a52c5942626e3e2256b3d78975d2740e9facb1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128880 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27245}
2019-03-22 12:59:48 +01:00
"../test:field_trial",
Refactoring mock_transport to be used separately Bug: webrtc:11251 Change-Id: I0a494c34c8d5c458b4d9b1b3616ae360d04df0d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168980 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Tim Na <natim@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30584}
2020-02-20 11:53:13 -08:00
"../test:mock_transport",
Fix deps of audio:audio_tests. Bug: webrtc:6828 Change-Id: Iae9020fda37fe40221d9a9def38c3afcc387d359 Reviewed-on: https://webrtc-review.googlesource.com/22683 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20666}
2017-11-13 17:04:05 +01:00
"../test:rtp_test_utils",
Passed AudioMixer to AudioState::Config. This is a refactoring change in preparation for enabling AudioMixer with the goal to have a small CL as possible for passing audio through the new audio mixer in WebRTC. The dependent CL https://codereview.webrtc.org/2436033002/ enables the mixer. An object of class AudioState is shared across different webrtc audio connections. It is created in tests and in WebRTCVoiceEngine. AudioState is constructed by passing a Config struct, where one argument is scoped_refptr<AudioMixer>. Populating this field has now been mandatory. Tests and WebRTCVoiceEngine create and pass either a AudioMixerImpl. WebRTCVoiceEngine passes a real AudioMixer, which is currently unused. An alternative would have tests pass a mocked audio mixer. We chose not to do that, because we believe that tests should use the real thing unless there are reasons against it. Construction time is not an issue, because the real mixer is relatively lightweight. We couldn't find a way to test any mixer-related changes in AudioState before the mixes is connected. The next dependent CL https://codereview.webrtc.org/2436033002/ contains unit tests for mixer usage. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2469743002 Cr-Commit-Position: refs/heads/master@{#15134}
2016-11-17 06:48:48 -08:00
"../test:test_common",
Move functionality out from AudioFrame and into AudioFrameOperations. This CL is in preparation to move the AudioFrame into webrtc/api. The AudioFrame is a POD type used for representing 10ms of audio. It appears as a parameter and return value of interfaces being migrated to webrtc/api, in particular AudioMixer. Here, methods operator+=, operator>>=, Mute are moved into a new target webrtc/audio/utility/audio_frame_operations, and dependencies are changed to use the new versions. The old AudioFrame methods are marked deprecated. The audio frame utilities in webrtc/modules/utility:audio_frame_operations are also moved to the new location. TBR=kjellander@webrtc.org BUG=webrtc:6548 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2424173003 Cr-Commit-Position: refs/heads/master@{#15413}
2016-12-05 01:46:09 -08:00
"../test:test_support",
GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries. BUG=webrtc:6954 R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2614833004 Cr-Commit-Position: refs/heads/master@{#15917}
2017-01-05 06:03:24 -08:00
"utility:utility_tests",
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
"//testing/gtest",
]
}
Adding placeholder for low bandwidth audio test This will allow the trybots to be updated to start running this new test executable, so that they can be used when landing this CL which will replace the dummy test with real code: https://codereview.webrtc.org/2694203002 Most likely, the trybots will just run the test binary while the perf bots will run a Python wrapper script that takes care of the post-processing to calculate audio quality using PESQ. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2717683002 Cr-Commit-Position: refs/heads/master@{#17063}
2017-03-06 04:01:16 -08:00
if (rtc_enable_protobuf) {
Add Windows, Mac, Android support to low bandwidth audio test BUG=webrtc:7229 Review-Url: https://codereview.webrtc.org/2767383005 Cr-Commit-Position: refs/heads/master@{#17470}
2017-03-30 04:01:30 -07:00
rtc_test("low_bandwidth_audio_test") {
Adding placeholder for low bandwidth audio test This will allow the trybots to be updated to start running this new test executable, so that they can be used when landing this CL which will replace the dummy test with real code: https://codereview.webrtc.org/2694203002 Most likely, the trybots will just run the test binary while the perf bots will run a Python wrapper script that takes care of the post-processing to calculate audio quality using PESQ. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2717683002 Cr-Commit-Position: refs/heads/master@{#17063}
2017-03-06 04:01:16 -08:00
testonly = true
sources = [
"test/low_bandwidth_audio_test.cc",
Reland "Fix collection of audio metrics from PC test framework for audio test" This is a reland of 2d0880b56954f57548deea51dfa678b80dbf618f To fix perf bot issue reading of perf results file was updated. Now perf results file will be generated by each test and then returned via output to the python script, which will get it and put into final file. Original change's description: > Fix collection of audio metrics from PC test framework for audio test > > Bug: webrtc:10138 > Change-Id: I18a8509a0cdc4ed1db6894c7540d5c0a155d6233 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144784 > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28514} Bug: webrtc:10138 Change-Id: I1347f09427736362a2d550612b48e09c06cfb1d1 No-Presubmit: True Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145201 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28527}
2019-07-10 14:40:58 +02:00
"test/low_bandwidth_audio_test_flags.cc",
Introduce PC level audio quality test. Add PC level audio quality test as separate file but into the same binary as low_bandwidth_audio_test.cc Bug: webrtc:10138 Change-Id: I3bb7df4130ab62c21a82881cd9f57b936f3cc11d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144621 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28493}
2019-07-05 12:34:36 +02:00
"test/pc_low_bandwidth_audio_test.cc",
Adding placeholder for low bandwidth audio test This will allow the trybots to be updated to start running this new test executable, so that they can be used when landing this CL which will replace the dummy test with real code: https://codereview.webrtc.org/2694203002 Most likely, the trybots will just run the test binary while the perf bots will run a Python wrapper script that takes care of the post-processing to calculate audio quality using PESQ. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2717683002 Cr-Commit-Position: refs/heads/master@{#17063}
2017-03-06 04:01:16 -08:00
]
Low-bandwidth audio testing The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout. The test cases are meant to try different network and audio configs (more are planned in the future). The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2694203002 Cr-Commit-Position: refs/heads/master@{#17356}
2017-03-23 03:40:03 -07:00
deps = [
- Removes voe_conference_test. - Adds a new AudioStatsTest, with better coverage of the same features, based on call_test. - Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses. BUG=webrtc:4690 R=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/3008273002 . Cr-Commit-Position: refs/heads/master@{#19833}
2017-09-14 14:46:47 +02:00
":audio_end_to_end_test",
Introduce PC level audio quality test. Add PC level audio quality test as separate file but into the same binary as low_bandwidth_audio_test.cc Bug: webrtc:10138 Change-Id: I3bb7df4130ab62c21a82881cd9f57b936f3cc11d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144621 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28493}
2019-07-05 12:34:36 +02:00
"../api:create_network_emulation_manager",
"../api:create_peerconnection_quality_test_fixture",
"../api:network_emulation_manager_api",
"../api:peer_connection_quality_test_fixture_api",
Introduce SimulatedNetworkReceiverInterface. Introduce SimulatedNetworkReceiverInterface and switch DirectTransport on this interface. Also switch part of related users on DefaultNetworkSimulationConfig. This two changes united into single CL to prevent work duplication. Most changes were done because of stop including fake_network_pipe.h into direct_transport.h, so splitting this into 2 CLs will require first fix all imports of fake_network_pipe.h and then replace them on new API imports again. Bug: webrtc:9630 Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6 Reviewed-on: https://webrtc-review.googlesource.com/94762 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24336}
2018-08-17 14:26:54 +02:00
"../api:simulated_network_api",
Introduce PC level audio quality test. Add PC level audio quality test as separate file but into the same binary as low_bandwidth_audio_test.cc Bug: webrtc:10138 Change-Id: I3bb7df4130ab62c21a82881cd9f57b936f3cc11d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144621 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28493}
2019-07-05 12:34:36 +02:00
"../call:simulated_network",
Low-bandwidth audio testing The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout. The test cases are meant to try different network and audio configs (more are planned in the future). The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2694203002 Cr-Commit-Position: refs/heads/master@{#17356}
2017-03-23 03:40:03 -07:00
"../common_audio",
"../system_wrappers",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../test:fileutils",
Reland "Fix collection of audio metrics from PC test framework for audio test" This is a reland of 2d0880b56954f57548deea51dfa678b80dbf618f To fix perf bot issue reading of perf results file was updated. Now perf results file will be generated by each test and then returned via output to the python script, which will get it and put into final file. Original change's description: > Fix collection of audio metrics from PC test framework for audio test > > Bug: webrtc:10138 > Change-Id: I18a8509a0cdc4ed1db6894c7540d5c0a155d6233 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144784 > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28514} Bug: webrtc:10138 Change-Id: I1347f09427736362a2d550612b48e09c06cfb1d1 No-Presubmit: True Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145201 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28527}
2019-07-10 14:40:58 +02:00
"../test:perf_test",
Low-bandwidth audio testing The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout. The test cases are meant to try different network and audio configs (more are planned in the future). The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2694203002 Cr-Commit-Position: refs/heads/master@{#17356}
2017-03-23 03:40:03 -07:00
"../test:test_common",
"../test:test_main",
Introduce PC level audio quality test. Add PC level audio quality test as separate file but into the same binary as low_bandwidth_audio_test.cc Bug: webrtc:10138 Change-Id: I3bb7df4130ab62c21a82881cd9f57b936f3cc11d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144621 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28493}
2019-07-05 12:34:36 +02:00
"../test:test_support",
"../test/pc/e2e:network_quality_metrics_reporter",
Reimplemeted "Test and fix for huge bwe drop after alr state" BUG=webrtc:7746 Test and fix for huge bwe drop after alr state. BUG=webrtc:7746 Review-Url: https://codereview.webrtc.org/2931873002 Cr-Commit-Position: refs/heads/master@{#18692} Committed: https://chromium.googlesource.com/external/webrtc/+/37aa8ba61641962119071646175bfe3bc2bda063 patch from issue 2931873002 at patchset 320001 (http://crrev.com/2931873002#ps320001) Review-Url: https://codereview.webrtc.org/2970653004 Cr-Commit-Position: refs/heads/master@{#19055}
2017-07-17 01:41:41 -07:00
"//testing/gtest",
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
"//third_party/abseil-cpp/absl/flags:flag",
Low-bandwidth audio testing The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout. The test cases are meant to try different network and audio configs (more are planned in the future). The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2694203002 Cr-Commit-Position: refs/heads/master@{#17356}
2017-03-23 03:40:03 -07:00
]
Adding placeholder for low bandwidth audio test This will allow the trybots to be updated to start running this new test executable, so that they can be used when landing this CL which will replace the dummy test with real code: https://codereview.webrtc.org/2694203002 Most likely, the trybots will just run the test binary while the perf bots will run a Python wrapper script that takes care of the post-processing to calculate audio quality using PESQ. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2717683002 Cr-Commit-Position: refs/heads/master@{#17063}
2017-03-06 04:01:16 -08:00
if (is_android) {
Low-bandwidth audio testing The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout. The test cases are meant to try different network and audio configs (more are planned in the future). The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2694203002 Cr-Commit-Position: refs/heads/master@{#17356}
2017-03-23 03:40:03 -07:00
deps += [ "//testing/android/native_test:native_test_native_code" ]
}
data = [
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
"../resources/voice_engine/audio_tiny16.wav",
Fix low_bandwidth_audio_perf_test resource dependency on Android The executable that's pushed to the device must depend on all files that need to be on the device. No-Try: True TBR: phoglund@webrtc.org Bug: chromium:755660 Change-Id: Iee041bd51e789e3ce6612fbda1582286a5cf4680 Reviewed-on: https://webrtc-review.googlesource.com/62961 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22494}
2018-03-19 15:58:08 +01:00
"../resources/voice_engine/audio_tiny48.wav",
Reland "Split perf-test-specific resources in low_bandwidth_audio_test" This is a reland of 4bbc150b18e961811991e3e524378e703b6d5b31 Now with explicitly specified `write_runtime_deps` Original change's description: > Split perf-test-specific resources in low_bandwidth_audio_test > > Bug: chromium:755660 > Change-Id: I7c60a47b26ad86892218497f28a09a04574077e6 > Reviewed-on: https://webrtc-review.googlesource.com/61961 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22439} Bug: chromium:755660 TBR: phoglund@webrtc.org Change-Id: I3d4bcc5156ee25de399ab23773ecb73cd995075c Reviewed-on: https://webrtc-review.googlesource.com/62660 Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22483}
2018-03-18 22:31:06 +01:00
]
}
group("low_bandwidth_audio_perf_test") {
testonly = true
Reland: Add in missing protobuf code. Turns out the import of histogram_pb2 failed not on the stub itself (which I thought for a long, long time), but because of the protobuf support code it includes in turn. This is a drawback of catching the ImportError in histogram_proto.py. This has a decent chance of fixing the problem. Tbr: mbonadei@webrtc.org Bug: chromium:1029452 Change-Id: If7ae2439b01ad1b3129d8cc8b158385101082e6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171867 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30917}
2020-03-27 09:11:30 +01:00
deps = [
":low_bandwidth_audio_test",
"//third_party/catapult/tracing/tracing/proto:histogram_proto",
"//third_party/protobuf:py_proto_runtime",
]
Reland "Split perf-test-specific resources in low_bandwidth_audio_test" This is a reland of 4bbc150b18e961811991e3e524378e703b6d5b31 Now with explicitly specified `write_runtime_deps` Original change's description: > Split perf-test-specific resources in low_bandwidth_audio_test > > Bug: chromium:755660 > Change-Id: I7c60a47b26ad86892218497f28a09a04574077e6 > Reviewed-on: https://webrtc-review.googlesource.com/61961 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22439} Bug: chromium:755660 TBR: phoglund@webrtc.org Change-Id: I3d4bcc5156ee25de399ab23773ecb73cd995075c Reviewed-on: https://webrtc-review.googlesource.com/62660 Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22483}
2018-03-18 22:31:06 +01:00
data = [
MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. NOTRY=True Bug: chromium:755660 Change-Id: I92de99cd1e3dd206f6cd366dbfd1c8c211d37cc7 Reviewed-on: https://webrtc-review.googlesource.com/4420 Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20089}
2017-09-28 16:14:37 +02:00
"test/low_bandwidth_audio_test.py",
Reland "Split perf-test-specific resources in low_bandwidth_audio_test" This is a reland of 4bbc150b18e961811991e3e524378e703b6d5b31 Now with explicitly specified `write_runtime_deps` Original change's description: > Split perf-test-specific resources in low_bandwidth_audio_test > > Bug: chromium:755660 > Change-Id: I7c60a47b26ad86892218497f28a09a04574077e6 > Reviewed-on: https://webrtc-review.googlesource.com/61961 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22439} Bug: chromium:755660 TBR: phoglund@webrtc.org Change-Id: I3d4bcc5156ee25de399ab23773ecb73cd995075c Reviewed-on: https://webrtc-review.googlesource.com/62660 Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22483}
2018-03-18 22:31:06 +01:00
"../resources/voice_engine/audio_tiny16.wav",
"../resources/voice_engine/audio_tiny48.wav",
Reland: Add in missing protobuf code. Turns out the import of histogram_pb2 failed not on the stub itself (which I thought for a long, long time), but because of the protobuf support code it includes in turn. This is a drawback of catching the ImportError in histogram_proto.py. This has a decent chance of fixing the problem. Tbr: mbonadei@webrtc.org Bug: chromium:1029452 Change-Id: If7ae2439b01ad1b3129d8cc8b158385101082e6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171867 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30917}
2020-03-27 09:11:30 +01:00
"${root_out_dir}/pyproto/tracing/tracing/proto/histogram_pb2.py",
Low-bandwidth audio testing The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout. The test cases are meant to try different network and audio configs (more are planned in the future). The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2694203002 Cr-Commit-Position: refs/heads/master@{#17356}
2017-03-23 03:40:03 -07:00
]
Reland "Switch low bw audio test to histograms." The reland makes sure the relevant code gets pulled by the isolate. Also requires a recipe change so the results processor switches to histogram mode when this CL is landed (see Chromium change 2119683). Bug: chromium:1029452 Change-Id: I18bc9de72c8d21cb2942ca9af774d3227a8bf874 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171693 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30894}
2020-03-26 08:41:09 +01:00
# TODO(http://crbug.com/1029452): Create a cleaner target with just the
# tracing python code. We don't need Polymer for instance.
data_deps = [ "//third_party/catapult/tracing:convert_chart_json" ]
Fix low_bandwidth_audio_perf_test binary dependency on Windows The split in https://webrtc-review.googlesource.com/c/src/+/62660 broke it. No-Try: True Bug: chromium:755660 Change-Id: I664f022cac9f8e7e0bb64a7cb59992f030543aa6 Reviewed-on: https://webrtc-review.googlesource.com/62801 Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22486}
2018-03-19 11:33:03 +01:00
if (is_win) {
data += [ "${root_out_dir}/low_bandwidth_audio_test.exe" ]
} else {
data += [ "${root_out_dir}/low_bandwidth_audio_test" ]
}
Make it possible to run low_bandwidth_audio_test on Android swarming. R=phoglund@webrtc.org, solenberg@webrtc.org Bug: chromium:755660 Change-Id: I8755a9c9df92fe8157c870cc7519130291441b25 Reviewed-on: https://webrtc-review.googlesource.com/39780 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21634}
2018-01-15 17:37:04 +01:00
if (is_linux || is_android) {
MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. NOTRY=True Bug: chromium:755660 Change-Id: I92de99cd1e3dd206f6cd366dbfd1c8c211d37cc7 Reviewed-on: https://webrtc-review.googlesource.com/4420 Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20089}
2017-09-28 16:14:37 +02:00
data += [
"../tools_webrtc/audio_quality/linux/PolqaOem64",
"../tools_webrtc/audio_quality/linux/pesq",
]
}
if (is_win) {
data += [
"../tools_webrtc/audio_quality/win/PolqaOem64.dll",
"../tools_webrtc/audio_quality/win/PolqaOem64.exe",
"../tools_webrtc/audio_quality/win/pesq.exe",
"../tools_webrtc/audio_quality/win/vcomp120.dll",
]
}
if (is_mac) {
data += [ "../tools_webrtc/audio_quality/mac/pesq" ]
}
Revert "Reland "Split perf-test-specific resources in low_bandwidth_audio_test"" This reverts commit b3808dcc36e4dca8b3d2b68c79e20c5888397690. Reason for revert: Still fails to generate runtime_deps Original change's description: > Reland "Split perf-test-specific resources in low_bandwidth_audio_test" > > This is a reland of 4bbc150b18e961811991e3e524378e703b6d5b31 > > Now using rtc_source_set to be able to generate runtime deps > > Original change's description: > > Split perf-test-specific resources in low_bandwidth_audio_test > > > > Bug: chromium:755660 > > Change-Id: I7c60a47b26ad86892218497f28a09a04574077e6 > > Reviewed-on: https://webrtc-review.googlesource.com/61961 > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22439} > > No-Try: True > Bug: chromium:755660 > Change-Id: I66eda6f016c98e2a8a99f230d9e0323cc09e4976 > Reviewed-on: https://webrtc-review.googlesource.com/62020 > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22450} TBR=phoglund@webrtc.org,oprypin@webrtc.org Change-Id: I781e3172416164e6d313574a31e4c982de8bcd9c No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: chromium:755660 Reviewed-on: https://webrtc-review.googlesource.com/62120 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22454}
2018-03-15 13:52:41 +00:00
Reland "Split perf-test-specific resources in low_bandwidth_audio_test" This is a reland of 4bbc150b18e961811991e3e524378e703b6d5b31 Now with explicitly specified `write_runtime_deps` Original change's description: > Split perf-test-specific resources in low_bandwidth_audio_test > > Bug: chromium:755660 > Change-Id: I7c60a47b26ad86892218497f28a09a04574077e6 > Reviewed-on: https://webrtc-review.googlesource.com/61961 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22439} Bug: chromium:755660 TBR: phoglund@webrtc.org Change-Id: I3d4bcc5156ee25de399ab23773ecb73cd995075c Reviewed-on: https://webrtc-review.googlesource.com/62660 Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22483}
2018-03-18 22:31:06 +01:00
write_runtime_deps = "${root_out_dir}/${target_name}.runtime_deps"
Adding placeholder for low bandwidth audio test This will allow the trybots to be updated to start running this new test executable, so that they can be used when landing this CL which will replace the dummy test with real code: https://codereview.webrtc.org/2694203002 Most likely, the trybots will just run the test binary while the perf bots will run a Python wrapper script that takes care of the post-processing to calculate audio quality using PESQ. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2717683002 Cr-Commit-Position: refs/heads/master@{#17063}
2017-03-06 04:01:16 -08:00
}
}
Reimplemeted "Test and fix for huge bwe drop after alr state" BUG=webrtc:7746 Test and fix for huge bwe drop after alr state. BUG=webrtc:7746 Review-Url: https://codereview.webrtc.org/2931873002 Cr-Commit-Position: refs/heads/master@{#18692} Committed: https://chromium.googlesource.com/external/webrtc/+/37aa8ba61641962119071646175bfe3bc2bda063 patch from issue 2931873002 at patchset 320001 (http://crrev.com/2931873002#ps320001) Review-Url: https://codereview.webrtc.org/2970653004 Cr-Commit-Position: refs/heads/master@{#19055}
2017-07-17 01:41:41 -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_perf_tests") {
Reimplemeted "Test and fix for huge bwe drop after alr state" BUG=webrtc:7746 Test and fix for huge bwe drop after alr state. BUG=webrtc:7746 Review-Url: https://codereview.webrtc.org/2931873002 Cr-Commit-Position: refs/heads/master@{#18692} Committed: https://chromium.googlesource.com/external/webrtc/+/37aa8ba61641962119071646175bfe3bc2bda063 patch from issue 2931873002 at patchset 320001 (http://crrev.com/2931873002#ps320001) Review-Url: https://codereview.webrtc.org/2970653004 Cr-Commit-Position: refs/heads/master@{#19055}
2017-07-17 01:41:41 -07:00
testonly = true
sources = [
"test/audio_bwe_integration_test.cc",
"test/audio_bwe_integration_test.h",
]
deps = [
Introduce SimulatedNetworkReceiverInterface. Introduce SimulatedNetworkReceiverInterface and switch DirectTransport on this interface. Also switch part of related users on DefaultNetworkSimulationConfig. This two changes united into single CL to prevent work duplication. Most changes were done because of stop including fake_network_pipe.h into direct_transport.h, so splitting this into 2 CLs will require first fix all imports of fake_network_pipe.h and then replace them on new API imports again. Bug: webrtc:9630 Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6 Reviewed-on: https://webrtc-review.googlesource.com/94762 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24336}
2018-08-17 14:26:54 +02:00
"../api:simulated_network_api",
Reland "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" This reverts commit 42d8c93ec351b68554825b58a3dc6525a7dc84da. Reason for revert: Got Aliby for FEC test flakes Original change's description: > Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" > > This reverts commit 304e9d2df347630d71fd4423f5971f30dac73e41. > > Reason for revert: Breaks downstream projects. > Seems to make VideoSendStreamTest.SupportsFlexfecSimulcastVp8 flaky. > > Original change's description: > > Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current > > > > Bug: webrtc:10191 > > Change-Id: I506cc50a90c73a6a4f6a3de36de0999cca72f5ba > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126230 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#27035} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org > > Change-Id: If98324f88e4b3d18bf2fe33597dfb9711867c243 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10191 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126484 > Reviewed-by: Yves Gerey <yvesg@webrtc.org> > Commit-Queue: Yves Gerey <yvesg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27041} TBR=danilchap@webrtc.org,kwiberg@webrtc.org,yvesg@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10191 Change-Id: Id87a17ae415142b8e0b11ba03ae7bad84a473fb0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126720 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Yves Gerey <yvesg@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27056}
2019-03-11 10:28:05 +00:00
"../api/task_queue",
Mark DirectTransport subclasses ctors as deprecated and switch from them Bug: webrtc:9630 Change-Id: I6e7bf898fd95ef76758458e759d9f9aa381f89e1 Reviewed-on: https://webrtc-review.googlesource.com/94843 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24345}
2018-08-20 13:30:39 +02:00
"../call:fake_network",
"../call:simulated_network",
Reimplemeted "Test and fix for huge bwe drop after alr state" BUG=webrtc:7746 Test and fix for huge bwe drop after alr state. BUG=webrtc:7746 Review-Url: https://codereview.webrtc.org/2931873002 Cr-Commit-Position: refs/heads/master@{#18692} Committed: https://chromium.googlesource.com/external/webrtc/+/37aa8ba61641962119071646175bfe3bc2bda063 patch from issue 2931873002 at patchset 320001 (http://crrev.com/2931873002#ps320001) Review-Url: https://codereview.webrtc.org/2970653004 Cr-Commit-Position: refs/heads/master@{#19055}
2017-07-17 01:41:41 -07:00
"../common_audio",
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 global task queue factory in audio/ unittests in particular replace rtc::TaskQueue with TaskQueueForTest class since latter uses DefaultTaskQueueFactory() directly instead of through GlobalTaskQueueFactory Bug: webrtc:10284 Change-Id: I1a52c5942626e3e2256b3d78975d2740e9facb1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128880 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27245}
2019-03-22 12:59:48 +01:00
"../rtc_base:task_queue_for_test",
Reimplemeted "Test and fix for huge bwe drop after alr state" BUG=webrtc:7746 Test and fix for huge bwe drop after alr state. BUG=webrtc:7746 Review-Url: https://codereview.webrtc.org/2931873002 Cr-Commit-Position: refs/heads/master@{#18692} Committed: https://chromium.googlesource.com/external/webrtc/+/37aa8ba61641962119071646175bfe3bc2bda063 patch from issue 2931873002 at patchset 320001 (http://crrev.com/2931873002#ps320001) Review-Url: https://codereview.webrtc.org/2970653004 Cr-Commit-Position: refs/heads/master@{#19055}
2017-07-17 01:41:41 -07:00
"../system_wrappers",
"../test:field_trial",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../test:fileutils",
Reimplemeted "Test and fix for huge bwe drop after alr state" BUG=webrtc:7746 Test and fix for huge bwe drop after alr state. BUG=webrtc:7746 Review-Url: https://codereview.webrtc.org/2931873002 Cr-Commit-Position: refs/heads/master@{#18692} Committed: https://chromium.googlesource.com/external/webrtc/+/37aa8ba61641962119071646175bfe3bc2bda063 patch from issue 2931873002 at patchset 320001 (http://crrev.com/2931873002#ps320001) Review-Url: https://codereview.webrtc.org/2970653004 Cr-Commit-Position: refs/heads/master@{#19055}
2017-07-17 01:41:41 -07:00
"../test:test_common",
"../test:test_main",
Harmonize paths to dependent targets. This CL consistently use: * relative paths for WebRTC dependent targets (test_support) * absolute paths for shared dependent targets (abseil) This is a necessary (but insufficient) step to build WebRTC tests from Chromium tree (rtc_include_tests=true), since test/ doesn't sit anymore in the top level directory. We also make sure that target declarations and uses are consistent in regard to build_with_chromium flag. Bug: webrtc:9943 Bug: webrtc:9855 Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b Reviewed-on: https://webrtc-review.googlesource.com/c/108720 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25445}
2018-10-30 21:12:42 +01:00
"../test:test_support",
Reimplemeted "Test and fix for huge bwe drop after alr state" BUG=webrtc:7746 Test and fix for huge bwe drop after alr state. BUG=webrtc:7746 Review-Url: https://codereview.webrtc.org/2931873002 Cr-Commit-Position: refs/heads/master@{#18692} Committed: https://chromium.googlesource.com/external/webrtc/+/37aa8ba61641962119071646175bfe3bc2bda063 patch from issue 2931873002 at patchset 320001 (http://crrev.com/2931873002#ps320001) Review-Url: https://codereview.webrtc.org/2970653004 Cr-Commit-Position: refs/heads/master@{#19055}
2017-07-17 01:41:41 -07:00
"//testing/gtest",
]
Reformat GN files. `gn format` recently [1] changed its formatting behavior for deps, source, and a few other elements when they are assigned (with =) single-element lists to be consistent with the formatting of updates (with +=) with single-element. Now that we've rolled in a GN binary with the change, reformat all files so that people don't get presubmit warnings due to this. CL generated with: $ git ls-files | grep BUILD.gn | xargs gn format $ gn format build_overrides/build.gni $ gn format build_overrides/gtest.gni $ gn format modules/audio_coding/audio_coding.gni $ gn format webrtc.gni $ gn format .gn Plus a few manual changes to add exceptions for "public_deps" (after changing these lines the presubmit started to complain). [1] - https://gn-review.googlesource.com/c/gn/+/6860 Bug: webrtc:11302 Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:10:10 +01:00
data = [ "//resources/voice_engine/audio_dtx16.wav" ]
Reimplemeted "Test and fix for huge bwe drop after alr state" BUG=webrtc:7746 Test and fix for huge bwe drop after alr state. BUG=webrtc:7746 Review-Url: https://codereview.webrtc.org/2931873002 Cr-Commit-Position: refs/heads/master@{#18692} Committed: https://chromium.googlesource.com/external/webrtc/+/37aa8ba61641962119071646175bfe3bc2bda063 patch from issue 2931873002 at patchset 320001 (http://crrev.com/2931873002#ps320001) Review-Url: https://codereview.webrtc.org/2970653004 Cr-Commit-Position: refs/heads/master@{#19055}
2017-07-17 01:41:41 -07:00
}
GN: Add video_engine_tests Adds separate source_sets for the video_engine_tests subtargets inside audio, call and video and merges them together into video_engine_tests. BUG=webrtc:5949 R=kjellander@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/2064523002 . Cr-Commit-Position: refs/heads/master@{#13127}
2016-06-14 12:52:54 +02:00
}
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