文件最后提交记录最后更新时间
Remove low_bandwidth_audio_test. Change-Id: Ide4d34e1dada9dc1448f89a79cc7b803ea4b5f46 Bug: b/284448060 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307160 Reviewed-by: Henrik Lundin <hlundin@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40191} 2 年前
Remove dependency on rtc_base_approved from most targets Bug: webrtc:9838 Change-Id: Ibd0199803597eff48ca139a5cecdc3209c62c5d2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259873 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36643} 4 年前
audio: fix some typos Bug: None Change-Id: I255a23a893d008dc58c3c9cb3facf61419c88c72 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320620 Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40779} 2 年前
Remove low_bandwidth_audio_test. Change-Id: Ide4d34e1dada9dc1448f89a79cc7b803ea4b5f46 Bug: b/284448060 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307160 Reviewed-by: Henrik Lundin <hlundin@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40191} 2 年前
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627} 3 年前
Add alessiob@webrtc.org in audio/OWNERS Bug: webrtc:10739 Change-Id: Iae658d7cd286c00f7065fce0681b0a61cd31f53b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274700 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38040} 3 年前
Migrate audio/ to use webrtc::Mutex Bug: webrtc:11567 Change-Id: Ic6a753f09aafb508690f4b8dadd4c99433fcfeb6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176741 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31635} 5 年前
Migrate audio/ to use webrtc::Mutex Bug: webrtc:11567 Change-Id: Ic6a753f09aafb508690f4b8dadd4c99433fcfeb6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176741 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31635} 5 年前
Propagate time of the last received packet with Timestamp type Bug: webrtc:13757 Change-Id: I446fc10ad6a90ab9ecaac337b9f2ad4ccad37cbd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307020 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40211} 2 年前
Use SequenceChecker(SequenceChecker::kDetached) in a few places. This CL is partly a test to see if there's an impact on binary size: - Not a big difference for binaries (decrease): -776b to -4Kb - For libraries (libwebrtc.a) it actually increases the size: +40Kb Secondarily this CL is basically to introduce this pattern to the code base. In terms of LOC, this makes things slightly more compact. From: class Foo { public: Foo() { checker_.Detach(); } private: SequenceChecker checker_; }; To: class Foo { public: Foo() = default; private: SequenceChecker checker_{SequenceChecker::kDetached}; }; Bug: none Change-Id: I59fc34ccea10847e13455a349851ce9a0af458e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299020 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39664} 3 年前
[SourceTracker] Move state to the worker thread, remove mutex. This is in preparation of using the state that SourceTracker manages for more things than only getContributingSources. Audio levels reported via getStats(), aren't consistent with levels reported via getCS. Since more operations will be derived from the ST owned data, moving the management of it away from the audio thread, reduces the potential of contention. Bug: webrtc:14029, webrtc:7517, webrtc:15119 Change-Id: I553f7e473316a1c61eeb43ded905a18242a04424 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302280 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39943} 3 年前
Simplify handling rtcp messages in audio send channel Delete VoERtcpObserver proxy: pass BWE related message directly to transport controller pass ReportBlock directly to ChannelSend, assuming there will be single report block per source ssrc Bug: None Change-Id: I8378326bff1dc3c2736960166fc782ee822a9c12 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305224 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40081} 2 年前
Simplify handling rtcp messages in audio send channel Delete VoERtcpObserver proxy: pass BWE related message directly to transport controller pass ReportBlock directly to ChannelSend, assuming there will be single report block per source ssrc Bug: None Change-Id: I8378326bff1dc3c2736960166fc782ee822a9c12 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305224 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40081} 2 年前
Remove internal overrides using old SendRtp and SendRtcp interfaces. This CL takes away all usages except for Android code. Low-Coverage-Reason: Refactoring old code Bug: webrtc:15410 Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40554} 2 年前
Pass rtcp message to RtpTransportController through newer interface NetworkLinkRtcpObserver is similar to RtcpBandwidthObserver but pass time variables using unit types instead of raw integers. Bug: webrtc:13757 Change-Id: Iaa0bbe0b108620b3a24013c40e7d9004032e904d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305022 Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40087} 2 年前
Use SequenceChecker(SequenceChecker::kDetached) in a few places. This CL is partly a test to see if there's an impact on binary size: - Not a big difference for binaries (decrease): -776b to -4Kb - For libraries (libwebrtc.a) it actually increases the size: +40Kb Secondarily this CL is basically to introduce this pattern to the code base. In terms of LOC, this makes things slightly more compact. From: class Foo { public: Foo() { checker_.Detach(); } private: SequenceChecker checker_; }; To: class Foo { public: Foo() = default; private: SequenceChecker checker_{SequenceChecker::kDetached}; }; Bug: none Change-Id: I59fc34ccea10847e13455a349851ce9a0af458e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299020 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39664} 3 年前
Use SequenceChecker(SequenceChecker::kDetached) in a few places. This CL is partly a test to see if there's an impact on binary size: - Not a big difference for binaries (decrease): -776b to -4Kb - For libraries (libwebrtc.a) it actually increases the size: +40Kb Secondarily this CL is basically to introduce this pattern to the code base. In terms of LOC, this makes things slightly more compact. From: class Foo { public: Foo() { checker_.Detach(); } private: SequenceChecker checker_; }; To: class Foo { public: Foo() = default; private: SequenceChecker checker_{SequenceChecker::kDetached}; }; Bug: none Change-Id: I59fc34ccea10847e13455a349851ce9a0af458e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299020 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39664} 3 年前
Implement support for Chrome task origin tracing. #3.5/4 This CL migrates unit tests to the new TaskQueueBase interface. Bug: chromium:1416199 Change-Id: Ic15c694b28eb67450ac99fdd56754de1246a4d95 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295621 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39434} 3 年前
Make capture timestamp optional in ADM. This is to avoid using 0 as a default value. Also fix a bug in audio_device_buffer where the timestamp aligner used the wrong input timestamp. Bug: webrtc:13609 Change-Id: I00016e68ab50d052990c2b9f80aa1e2d7e167b93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291118 Reviewed-by: Olov Brändström <brandstrom@google.com> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39177} 3 年前
Make capture timestamp optional in ADM. This is to avoid using 0 as a default value. Also fix a bug in audio_device_buffer where the timestamp aligner used the wrong input timestamp. Bug: webrtc:13609 Change-Id: I00016e68ab50d052990c2b9f80aa1e2d7e167b93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291118 Reviewed-by: Olov Brändström <brandstrom@google.com> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39177} 3 年前
Change ChannelReceive::GetAudioFrameWithInfo to use new Converts method Use the new Converts function added in webrtc-review.googlesource.com/c/src/+/320080. Later this will also be added to video. This change is part of an effort to get Glass 2 Glass metrics. This particular change is not needed, but I intend to add this code to video, and thinks it's nice if the code for video and audio looks the same. Bug: None Change-Id: I04caff0dbef1cd4f391bbaa4f8bdee0e66043888 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320281 Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Commit-Queue: Olov Brändström <brandstrom@google.com> Cr-Commit-Position: refs/heads/main@{#40753} 2 年前
Propagate time of the last received packet with Timestamp type Bug: webrtc:13757 Change-Id: I446fc10ad6a90ab9ecaac337b9f2ad4ccad37cbd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307020 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40211} 2 年前
Support receiving cloned encoded audio frames Bug: chromium:1464860 Change-Id: I01b2d768fcf5aef09b32304a8f9fe0b00ca32357 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316320 Reviewed-by: Tony Herre <herre@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Palak Agarwal <agpalak@google.com> Cr-Commit-Position: refs/heads/main@{#40583} 2 年前
Use backticks not vertical bars to denote variables in comments for /audio Bug: webrtc:12338 Change-Id: Ief89269aa39d0cb6749a1c6cc995ce8830ca327f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226942 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34564} 4 年前
Support sending IncomingFrames in audio ChannelSendFrameTransformerDelegate::SendFrame() currently only supports sending frames in a single direction. With this change, we allow sending received audio frames. Bug: chromium:1464847 Change-Id: I8113a3278dfce7b2ba709afecc672bc9af9c4a27 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316600 Reviewed-by: Tony Herre <herre@google.com> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40643} 2 年前
Remove internal overrides using old SendRtp and SendRtcp interfaces. This CL takes away all usages except for Android code. Low-Coverage-Reason: Refactoring old code Bug: webrtc:15410 Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40554} 2 年前
Make field trial string DisableRtxRateLimiter enabled by default. Bug: webrtc:15184 Change-Id: Ie2a20892b71defe2a3b744ae5b631a76f9a8712c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325120 Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41016} 2 年前
Simplify handling rtcp messages in audio send channel Delete VoERtcpObserver proxy: pass BWE related message directly to transport controller pass ReportBlock directly to ChannelSend, assuming there will be single report block per source ssrc Bug: None Change-Id: I8378326bff1dc3c2736960166fc782ee822a9c12 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305224 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40081} 2 年前
Pass the correct abs_capture_timestamp while cloning audio frame This change replaces type of absolute_capture_timestamp_ms_ in TransformableOutgoingAudioFrame from int to optional uint and makes the function AbsoluteCaptureTimestamp() inside TransformableAudioFrameInterface pure virtual. Bug: webrtc:14949 Change-Id: Id3bdbcba63a5f91105ab198208e4f2b11eb3c7db Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319000 Commit-Queue: Palak Agarwal <agpalak@google.com> Reviewed-by: Tony Herre <herre@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40814} 2 年前
Move RTPTimestamp offset handling out of encoded transform delegate Keep the logic managing whether audio RTP timestamps have the random start offset added or not inside ChannelSend, so that the ChannelSendFrameTransformerDelegate doesn't need to worry about it. Crucially, this means that frames moved between senders by encoded transforms clients will always use the correct offset for the channel where we actually get sent. Also rename TS variables throughout both classes to be explicit over whether the offset has been added or not. Bug: chromium:1464847 Change-Id: I19955ec4c1cb834161b00dd74622725a070b713a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317900 Commit-Queue: Tony Herre <herre@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40655} 2 年前
Move RTPTimestamp offset handling out of encoded transform delegate Keep the logic managing whether audio RTP timestamps have the random start offset added or not inside ChannelSend, so that the ChannelSendFrameTransformerDelegate doesn't need to worry about it. Crucially, this means that frames moved between senders by encoded transforms clients will always use the correct offset for the channel where we actually get sent. Also rename TS variables throughout both classes to be explicit over whether the offset has been added or not. Bug: chromium:1464847 Change-Id: I19955ec4c1cb834161b00dd74622725a070b713a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317900 Commit-Queue: Tony Herre <herre@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40655} 2 年前
Move RTPTimestamp offset handling out of encoded transform delegate Keep the logic managing whether audio RTP timestamps have the random start offset added or not inside ChannelSend, so that the ChannelSendFrameTransformerDelegate doesn't need to worry about it. Crucially, this means that frames moved between senders by encoded transforms clients will always use the correct offset for the channel where we actually get sent. Also rename TS variables throughout both classes to be explicit over whether the offset has been added or not. Bug: chromium:1464847 Change-Id: I19955ec4c1cb834161b00dd74622725a070b713a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317900 Commit-Queue: Tony Herre <herre@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40655} 2 年前
Make header files self contained. This CL adds #includes to header files in order to make them self contained after the preprocessor pass. Bug: b/251890128 Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38327} 3 年前
Simplify handling rtcp messages in audio send channel Delete VoERtcpObserver proxy: pass BWE related message directly to transport controller pass ReportBlock directly to ChannelSend, assuming there will be single report block per source ssrc Bug: None Change-Id: I8378326bff1dc3c2736960166fc782ee822a9c12 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305224 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40081} 2 年前
Reland "Rename FATAL() into RTC_FATAL()." This is a reland of 9653d26f8e83bb685477e7ba5c2adf2863187743 Original change's description: > Rename FATAL() into RTC_FATAL(). > > No-Try: True > Bug: webrtc:8454 > Change-Id: I9130487a92463a2128cf1493e6c5117b2fab313a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193703 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32620} No-Try: True Bug: webrtc:8454 Change-Id: Idb80125ac31ea307d1434bc9a65f148ac2017a3c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193864 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32635} 5 年前
Use backticks not vertical bars to denote variables in comments for /audio Bug: webrtc:12338 Change-Id: Ief89269aa39d0cb6749a1c6cc995ce8830ca327f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226942 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34564} 4 年前
Clarify and extend test support for certain sample rates in audio processing Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly. This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better. This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API. Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000. Bug: chromium:1332484 Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37682} 3 年前