- 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
- bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 6 years ago
- abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 6 years ago
- 9c1ee36 Fix low_bandwidth_audio_perf_test resource dependency on Android by Oleh Prypin · 7 years ago
- 7b2676f Fix low_bandwidth_audio_perf_test binary dependency on Windows by Oleh Prypin · 7 years ago
- 8cf0a87 Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
- 7696bef Remove the public_deps to fileutils from test_support. by Patrik Höglund · 7 years ago
- 650a826 Revert "Reland "Split perf-test-specific resources in low_bandwidth_audio_test"" by Oleh Prypin · 7 years ago
- b3808dc Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
- aaa882c Revert "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
- 4bbc150 Split perf-test-specific resources in low_bandwidth_audio_test by Oleh Prypin · 7 years ago
- 3faa832 Separate test/fake_audio_device on API and implementation. Step 2. by Artem Titov · 7 years ago
- 12edf4c Separate build target for rtc_base/numerics/safe_minmax.h by Karl Wiberg · 7 years ago
- 6723cdc Revert "Separate test/fake_audio_device on API and implementation." by Artem Titov · 7 years ago
- 8ea5f9a Separate test/fake_audio_device on API and implementation. by Artem Titov · 7 years ago
- fef0500 Adding a new string utility class: SimpleStringBuilder. by Tommi · 7 years ago
- f35c666 Separate build targets for aec3 and aec3_unittests by Gustaf Ullberg · 7 years ago
- ef9daee Using mock transport controller in audio unit tests. by Sebastian Jansson · 7 years ago
- 1896cec Removed dependencies from audio send stream unit test by Sebastian Jansson · 7 years ago
- 2ae140a BUILD.gn file for api/audio. by Gustaf Ullberg · 7 years ago
- e4be6da Removing access to send side cc in rtp controller. by Sebastian Jansson · 7 years ago
- dbbb33c Stop using public_deps in common_audio. by Mirko Bonadei · 7 years ago
- 970b088 Reland "Break up rtc_event_log_api to solve circular dependencies." by Qingsi Wang · 7 years ago
- 75df728 Revert "Break up rtc_event_log_api to solve circular dependencies." by Mirko Bonadei · 7 years ago
- 001546d Break up rtc_event_log_api to solve circular dependencies. by Qingsi Wang · 7 years ago
- 65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
- a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
- 98d4036 Make it possible to run low_bandwidth_audio_test on Android swarming. by Edward Lemur · 7 years ago
- 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
- 731082c Reland: Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
- 5a25ab2 Revert "Add mock_rtc_event_log.h." by Edward Lemur · 7 years ago
- 63aea46 Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
- 94dc177 Add mock_bitrate_controller.h. by Patrik Höglund · 7 years ago
- 6213929 Add missing files to audio_processing. by Patrik Höglund · 7 years ago
- 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
- a8005cf Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
- d37709b Revert "Fix circular dependencies between optional, array_view, and rtc_base." by Patrik Höglund · 7 years ago
- a9e0924 Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
- cedd351 Do not add audio bitrate observer if TWCC sending is not supported by Alex Narest · 7 years ago
- b5728d9 Stop using public_deps in modules/rtp_rtcp. by Mirko Bonadei · 7 years ago
- 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
- c0e6804 Fix deps of audio:audio_tests. by Patrik Höglund · 7 years ago
- 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