- 7a9a092 Delete media transport integration. by Bjorn A Mellem · 5 years ago
- cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
- ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
- eb90e6f Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest by Danil Chapovalov · 5 years ago
- ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
- fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
- 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
- 40de3cc Propagating TargetRate struct to BitrateAllocator. by Sebastian Jansson · 5 years ago
- 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 5 years ago
- 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
- 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
- 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
- 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 5 years ago
- a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
- d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
- 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
- 9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 5 years ago
- 413ccc4 Stop DCHECK which occurs in ANA BitrateController when overhead is zero. by Bjorn A Mellem · 5 years ago
- 63658d0 Revert "Ensure that we always set values for min and max audio bitrate." by Daniel Lee · 6 years ago
- e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
- 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 6 years ago
- 31660fd Avoid using global task queue factory in audio/ unittests by Danil Chapovalov · 6 years ago
- 44dd9f2 Adds ChannelSend specific encoder task queue. by Sebastian Jansson · 6 years ago
- 0b69826 Don't inject worker queue into send streams. by Sebastian Jansson · 6 years ago
- ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 6 years ago
- 8fb1a6a Delete a few return values from audio streams and video send streams. by Niels Möller · 6 years ago
- 977b335 Injecting Clock into audio streams. by Sebastian Jansson · 6 years ago
- da6806c Injecting Clock into BitrateAllocator. by Sebastian Jansson · 6 years ago
- 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
- 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
- fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
- 14a7cf9 Adds CallEncoder to ChannelSend. by Sebastian Jansson · 6 years ago
- 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
- 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
- e7d08df Fix chromium roll into WebRTC. by Artem Titov · 6 years ago
- 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
- f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
- e977199 Delete ChannelSend::RegisterTransport, replacing by construction argument by Niels Möller · 6 years ago
- ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
- 254d869 Routing BitrateAllocationUpdate to audio codec. by Sebastian Jansson · 6 years ago
- 13e5903 Using unit classes in BitrateAllocationUpdate struct. by Sebastian Jansson · 6 years ago
- c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
- dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
- 645a3af Remove unused/unnecessary things from ChannelSend. by Fredrik Solenberg · 6 years ago
- 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
- 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
- 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
- 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
- 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
- c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
- 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
- 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
- b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
- 35fa280 Adds allocated rate without feedback to new congestion controller. by Sebastian Jansson · 6 years ago
- 9701e0c Makes treatment of received reports of packets lost signed. by Sebastian Jansson · 6 years ago
- fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
- 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
- 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
- 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
- b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
- 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
- 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 7 years ago
- fe617a3 Adding has_packet_feedback to LimitObserver callback. by Sebastian Jansson · 7 years ago
- d6fbf2a Tests: Pass codec ID argument to audio codecs by Karl Wiberg · 7 years ago
- f69e768 Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 1. by philipel · 7 years ago
- ef9daee Using mock transport controller in audio unit tests. by Sebastian Jansson · 7 years ago
- 41f16be Silencing warnings in audio send stream unit tests. by Sebastian Jansson · 7 years ago
- 97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
- 1896cec Removed dependencies from audio send stream unit test by Sebastian Jansson · 7 years ago
- 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
- a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
- 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
- f85e31b Don't (re-)configure BitrateObserver unless already sending by Oskar Sundbom · 7 years ago
- 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
- 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
- d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
- 2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 7 years ago
- 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
- fc3a2e3 Remove the VoiceEngineObserver callback interface. 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/audio_send_stream_unittest.cc]
- e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
- 5c8942a Move PacedSender ownership to RtpTransportControllerSend. by Stefan Holmer · 7 years ago
- 8de1826 Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by minyue-webrtc · 7 years ago
- 7df370b Revert "Allow AudioSendStream to reconfig AudioNetworkAdaptor" by Minyue Li · 7 years ago
- 4a88120 Allow AudioSendStream to reconfig AudioNetworkAdaptor by minyue-webrtc · 7 years ago
- e76bd3a 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 · 7 years ago
- c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
- a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
- c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
- 1129df2 Always ResetSenderCongestionControlObjects before RegisterEtc... by ossu · 7 years ago
- a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
- c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 7 years ago
- 8c96a14 Simple tests for Call::SetBitrateConfig. by zstein · 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
- 3b9ff38 Have AudioSendStream register CNG payload types with the RtpRtcpModule. by ossu · 7 years ago
- 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
- cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 7 years ago
- fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 8 years ago