- 0c96449 Clamp stable target bitrate to min/max allocated bitrate. by Jakob Ivarsson · 4 years, 7 months ago
- c771084 Remove RTC_NOTREACHED from audio_send_stream when ANA didn't work by Alejandro Luebs · 4 years, 7 months ago
- 01ab084 Add minimum overhead to configured priorty bitrate instead of maximum. by Jakob Ivarsson · 4 years, 7 months ago
- d14525e Make sure that the audio stream is allocated with the correct overhead. by Jakob Ivarsson · 4 years, 7 months ago
- 74dadc1 Ready to support of absolute capture timestamp header extension. by Minyue Li · 4 years, 7 months ago
- cad3e0e Replace DataSize and DataRate factories with newer versions by Danil Chapovalov · 4 years, 8 months ago
- 0c626af Use newer version of TimeDelta and TimeStamp factories in webrtc by Danil Chapovalov · 4 years, 8 months ago
- bef818d Default disables legacy overhead calculation. by Sebastian Jansson · 4 years, 9 months ago
- c3eb9fd Reland "Reland "Only include overhead if using send side bandwidth estimation."" by Sebastian Jansson · 4 years, 9 months ago
- 4356490 Revert "Reland "Only include overhead if using send side bandwidth estimation."" by Mirko Bonadei · 4 years, 9 months ago
- 086055d Reland "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 4 years, 9 months ago
- c709412 Revert "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 4 years, 9 months ago
- 8c79c6e Only include overhead if using send side bandwidth estimation. by Sebastian Jansson · 4 years, 9 months ago
- 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 4 years, 9 months ago
- 7a9a092 Delete media transport integration. by Bjorn A Mellem · 4 years, 11 months ago
- cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 5 years ago
- 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
- f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 5 years ago
- ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 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
- cd0eedb Don't allocate audio if we have no transport sequence number. by Sebastian Jansson · 5 years ago
- 0a6510d Removes rtp_transport checks in AudioSendStream by Sebastian Jansson · 5 years ago
- 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 5 years ago
- 0429f78 Base overhead calculation for audio priority rate on available data. by Sebastian Jansson · 5 years ago
- f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 5 years ago
- 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
- 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
- 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
- f13df86 Delete audio methods SignalNetworkState by Niels Möller · 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
- 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 5 years ago
- 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 5 years ago
- 1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 5 years ago
- aa59eca Move RtpPacketSender and merge it with RtpPacketPacer. by Erik Språng · 5 years ago
- 0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 5 years ago
- 4c2c412 Set local ssrc at construction (audio) by Erik Språng · 5 years ago
- e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 5 years ago
- 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 5 years ago
- d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
- 6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
- a352248 Add a config flag to disable the audio ALR probing request. by Christoffer Rodbro · 5 years ago
- 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
- 8f119ca Enable experiments with audio bitrate priority. by Jonas Olsson · 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 · 5 years ago
- e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 5 years ago
- cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 5 years ago
- c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
- 741daaf Move rtc::FunctionView to the public API by Artem Titov · 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
- 8672cac Trigger audio bitrate allocation update on overhead change. by Sebastian Jansson · 6 years ago
- ee5ccbc Move ownership of RTPSenderAudio to ChannelSend. by Niels Möller · 6 years ago
- 110c64b Delete unused key WebRTC-Audio-SendSideBwe-For-Video. by Christoffer Rodbro · 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
- 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
- 464a557 Adds audio priority bitrate field trial parameter. by Sebastian Jansson · 6 years ago
- 626015d Make AudioSendStream to be OverheadObserver by Anton Sukhanov · 6 years ago
- 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
- 79f0d4d Enables feature to account for unacknowledged data. by Sebastian Jansson · 6 years ago
- 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