- 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 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
- 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 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
- 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
- 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
- c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
- 2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 6 years ago
- 56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 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
- 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
- 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
- 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
- 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
- bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
- b686396 Makes AudioSendStream signal that it's part of allocation. 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
- 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
- bbbe4e1 Better handle target audio bitrate allocation. by Alex Narest · 6 years ago
- bcf9180 Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream. by Alex Narest · 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
- 867e510 Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled. by Alex Narest · 6 years ago
- 24ad720 Uses config struct with bitrate allocator. by Sebastian Jansson · 6 years ago
- abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 6 years ago
- 003930a Fix MID not always getting set with audio by Steve Anton · 6 years ago
- bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 6 years ago
- 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 6 years ago
- 77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 6 years ago
- 763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 6 years ago
- 98cd810 Production code: Pass codec ID argument to audio codecs by Karl Wiberg · 6 years ago
- 3c24ea8 Removed SetTransportOverhead in transport controller. by Sebastian Jansson · 6 years ago
- 4c1ffb8 Removing access to pacer in rtp controller. by Sebastian Jansson · 7 years ago
- e4be6da Removing access to send side cc in rtp controller. by Sebastian Jansson · 7 years ago
- 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
- 24ea822 Remove logging in audio/* from release builds. by Jonas Olsson · 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
- 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
- 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
- d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
- f85e31b Don't (re-)configure BitrateObserver unless already sending by Oskar Sundbom · 7 years ago
- aaedf75 Replace VoEBase::[Start/Stop]Send(). by Fredrik Solenberg · 7 years ago
- 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
- cedd351 Do not add audio bitrate observer if TWCC sending is not supported by Alex Narest · 7 years ago
- 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
- 2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 7 years ago
- 8d9c540 Deprecated BitrateController::CreateRtcpBandwidthObserver. by Sebastian Jansson · 7 years ago
- 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
- 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
- dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
- a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
- 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
- 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
- b3944f0 Media track ID visibility at BWE level by Alex Narest · 7 years ago
- 90e1f53 Fix potentional race in AudioSendStream constructor by Danil Chapovalov · 7 years ago
- 1c239d4 Remove voe::Statistics. 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.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
- 3651fdd Uncomment thread-check in AudioSendStream::OnPacketFeedbackVector() by eladalon · 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
- abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
- c58f8c0 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 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
- edd6eea Rename elad.alon to eladalon, to avoid confusion between repositories. by eladalon · 7 years ago
- c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 7 years ago
- 93e4522 Renaming probing_interval to bwe_period globally. by minyue · 7 years ago
- 48368ad Fixing video loopback test with encoder factory. by minyue · 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
- fca900a Fix two invalid DCHECKs related to audio BWE. by stefan · 7 years ago
- fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 7 years ago
- 4e76451 Fix UT failure by temporarily uncommenting by elad.alon · 7 years ago
- b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 7 years ago
- dadb4dc Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 7 years ago
- d12a8e1 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 7 years ago
- 559af38 Split CongestionController into send- and receive-side classes. by nisse · 7 years ago
- 5bbf43f Move delay_based_bwe_ into CongestionController by elad.alon · 7 years ago