Log - 507f43465b3306f3ba1c3ff639e7a46ab33a8904 - platform/external/webrtc

61a7b14 Removing conditional visibility. by Mirko Bonadei · 7 years ago
5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
2011075 MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. by Edward Lemur · 7 years ago
18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 7 years ago
92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/BUILD.gn]
73276ad - Removes voe_conference_test. by Fredrik Solenberg · 7 years ago
84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
9b2f20c Replace gflags usages with rtc_base/flags in all targets based on test_main by oprypin · 7 years ago
413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
037f3e4 Replace absolute path with relative path for GN files. by Jianjun Zhu · 7 years ago
f6a861a Remove remains of webrtc/base by ehmaldonado · 7 years ago
c58f8c0 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 7 years ago
9d11764 Reimplemeted "Test and fix for huge bwe drop after alr state" by tschumim · 7 years ago
c024740 Use relative paths in GN files. by jianjun.zhu · 7 years ago
370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 7 years ago
9483b49 Remove remains of webrtc/base by ehmaldonado · 7 years ago
e75d96b Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ ) by terelius · 7 years ago
0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 7 years ago
37aa8ba Test and fix for huge bwe drop after alr state. by tschumim · 7 years ago
d76b7b2 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 7 years ago
7cb69d5 This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 7 years ago
eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
e0629c0 GN: Tighten up test target visibility + refactorings by kjellander · 7 years ago
f250100 Add POLQA to low bandwidth audio test by oprypin · 7 years ago
6d305ba Add Windows, Mac, Android support to low bandwidth audio test by oprypin · 7 years ago
92220ff Low-bandwidth audio testing by oprypin · 7 years ago
5e1ca78 Add low_bandwidth_audio_test to default build by oprypin · 8 years ago
8f8d1a0 Adding placeholder for low bandwidth audio test by kjellander · 8 years ago
7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
9aa3f0a Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) by mbonadei · 8 years ago
69dc7db Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) by mbonadei · 8 years ago
35a3270 Moving webrtc.gni up one level from build/ by mbonadei · 8 years ago
894c2bb GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries. by ehmaldonado · 8 years ago
676e08f Refactor webrtc/{api,audio} and modules/audio_coding for GN check by kjellander · 8 years ago
f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
6321b49 Move functionality out from AudioFrame and into AudioFrameOperations. by aleloi · 8 years ago
939e08f Added webrtc/audio/utility directory and empty GN target. by aleloi · 8 years ago
04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
aed581a Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
e40a7ee GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
b62dbbe GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 8 years ago
e9cc686 GN Templates: Move common_inherited_config to the template. by ehmaldonado · 8 years ago
7a2ce0b GN Templates: Move common_config to the template. by ehmaldonado · 8 years ago
38a2132 GN: Introduce templates. by ehmaldonado · 8 years ago
a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
0208322 GN: Add video_engine_tests by Peter Boström · 8 years ago
50772f1 GN: Update audio_sink.h location by kjellander@webrtc.org · 9 years ago
f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
43e83d4 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
a457752 Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
c7a8b08 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago
5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 9 years ago