- 642a074 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
- 3aa28f0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
- 5b18967 Move optional.h to webrtc/api/ by kwiberg · 7 years ago
- 00ed864 Recently we moved webrtc/base to webrtc/rtc_base, so these by mbonadei · 7 years ago
- 0a6becd Replace gflags usages with rtc_base/flags in all targets based on test_main by oprypin · 7 years ago
- 443f9a9 Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
- 94ac82f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
- 7a9414b Uncomment thread-check in AudioSendStream::OnPacketFeedbackVector() by eladalon · 7 years ago
- 06f3606 Replacing NetEq discard rate with secondary discarded rate. by minyue-webrtc · 7 years ago
- 53431d2 Move PacedSender ownership to RtpTransportControllerSend. by Stefan Holmer · 7 years ago
- 0f007ea Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
- de6c74c Replace absolute path with relative path for GN files. by Jianjun Zhu · 7 years ago
- 090170a Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by minyue-webrtc · 7 years ago
- 2af0153 Revert "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by Minyue Li · 7 years ago
- 3e17c5a Allow AudioSendStream to reconfig AudioNetworkAdaptor by minyue-webrtc · 8 years ago
- ea04cc0 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
- 2bc62f3 Remove remains of webrtc/base by ehmaldonado · 8 years ago
- a809196 Disable flaky NoBandwidthDropAfterDtx test. by tschumim · 8 years ago
- a42f080 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 8 years ago
- ea0d2c7 Don't run NoBandwidthDropAfterDtx test on andriod because it's flaky. by tschumim · 8 years ago
- 4098a82 Reimplemeted "Test and fix for huge bwe drop after alr state" by tschumim · 8 years ago
- d1701d0 Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. by zstein · 8 years ago
- 59d5575 Use relative paths in GN files. by jianjun.zhu · 8 years ago
- 4f870fc Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 8 years ago
- c6c814d Remove remains of webrtc/base by ehmaldonado · 8 years ago
- b7b8932 External APM usage downstream dependency support cleanup by peah · 8 years ago
- 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
- bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
- 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
- dbf8bcf Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ ) by terelius · 8 years ago
- 7213eea Always ResetSenderCongestionControlObjects before RegisterEtc... by ossu · 8 years ago
- 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
- 8699ef2 Disable AudioBweIntegrationTest.NoBandwidthDropAfterDtx - it's flaky by eladalon · 8 years ago
- 04ef6b0 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
- cb68c2d Test and fix for huge bwe drop after alr state. by tschumim · 8 years ago
- e666375 Also scan stderr for audio files to test, due to change in Android test_runner by oprypin · 8 years ago
- 7d97307 Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide: by yujo · 8 years ago
- 2d1de82 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 8 years ago
- b11cc61 Rename elad.alon to eladalon, to avoid confusion between repositories. by eladalon · 8 years ago
- 30fa632 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 8 years ago
- a4c5957 Renaming probing_interval to bwe_period globally. by minyue · 8 years ago
- 82158aa Simple tests for Call::SetBitrateConfig. by zstein · 8 years ago
- 78bbf3a New class RtpDemuxer and RtpPacketSinkInterface, use in Call. by nisse · 8 years ago
- b2a8855 Fixing video loopback test with encoder factory. by minyue · 8 years ago
- 4647b14 Drop deprecated AudioFrameOperations::Scale method signatures by oprypin · 8 years ago
- eda1919 Rename tools-webrtc -> tools_webrtc by Henrik Kjellander · 8 years ago
- 29a1f8c This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 8 years ago
- cfbbdcc Add --quick flag to low bandwidth audio test by oprypin · 8 years ago
- f26202b Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
- 30af12b Have AudioSendStream register CNG payload types with the RtpRtcpModule. by ossu · 8 years ago
- fe5f71c Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
- ce4e632 Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) by mbonadei · 8 years ago
- 0d80b56 Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) by mbonadei · 8 years ago
- b70f74c Creating webrtc/modules:module_api by mbonadei · 8 years ago
- 9cdb538 GN: Tighten up test target visibility + refactorings by kjellander · 8 years ago
- cddf701 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
- 3d14b94 Fix lint errors to enable stricter PyLint rules by kjellander · 8 years ago
- 35648bc Add POLQA to low bandwidth audio test by oprypin · 8 years ago
- b9f62aa Making FakeNetworkPipe demux audio and video packets. by minyue · 8 years ago
- 69579d7 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
- af8f6e5 Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
- fee642e Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
- d6bc12c Fix two invalid DCHECKs related to audio BWE. by stefan · 8 years ago
- b5eeeee Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
- 085ed30 Support multiple connected Android devices in low bandwidth audio test by oprypin · 8 years ago
- 5e9233b Let PacketRouter separate send and receive modules. by nisse · 8 years ago
- 7843bda Add Windows, Mac, Android support to low bandwidth audio test by oprypin · 8 years ago
- a54ea6e Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) by nisse · 8 years ago
- f67ccf5 Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) by lliuu · 8 years ago
- 9b9b7e2 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 8 years ago
- e153ae8 Fix UT failure by temporarily uncommenting by elad.alon · 8 years ago
- afd1255 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
- 22de8f3 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
- 52b9df6 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
- 135be28 WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
- 1993d91 Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 8 years ago
- 49d1987 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 8 years ago
- 1b0415d Low-bandwidth audio testing by oprypin · 8 years ago
- a58b581 Reland of Delete class MockCongestionController. (patchset #1 id:1 of https://codereview.webrtc.org/2762133003/ ) by nisse · 8 years ago
- 7a8eb23 Revert of Delete class MockCongestionController. (patchset #4 id:60001 of https://codereview.webrtc.org/2762023004/ ) by skvlad · 8 years ago
- b1d9258 Delete class MockCongestionController. by nisse · 8 years ago
- 4e519ca Split CongestionController into send- and receive-side classes. by nisse · 8 years ago
- 109f6f7 Add low_bandwidth_audio_test to default build by oprypin · 8 years ago
- 9268abe Move delay_based_bwe_ into CongestionController by elad.alon · 8 years ago
- 0f20d58 Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ ) by oprypin · 8 years ago
- 899c143 Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ ) by oprypin · 8 years ago
- ba76094 Enable cpplint and fix cpplint errors in webrtc/*audio by oprypin · 8 years ago
- 68e2137 Adding placeholder for low bandwidth audio test by kjellander · 8 years ago
- 5d07023 Remove MockRemoteBitrateObserver (unused) by elad.alon · 8 years ago
- bf77153 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
- f21613d Support 4 channel mic in Windows Core Audio by jens.nielsen · 8 years ago
- 9617a87 Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
- 95a13b4 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
- ac16f1b Remove VoEVideoSync interface. by solenberg · 8 years ago
- 660a2e0 Clean out platform specific things from voice_engine_defines.h. by solenberg · 8 years ago
- c10c31d Wire up audio packet loss to BWE. by stefan · 8 years ago
- 49cdd35 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
- f65fb4c Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
- 6045993 Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
- a0add70 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago