Log - 53e048d83aa47de7bf98d1f10c667bd0e79551e7 - platform/external/webrtc

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
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
8d9c540 Deprecated BitrateController::CreateRtcpBandwidthObserver. by Sebastian Jansson · 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.h]
a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 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
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
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
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
b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 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
7de8d64 Wire up audio packet loss to BWE. by stefan · 7 years ago
f4caaab Fix for bwe with overhead on audio only calls. by michaelt · 8 years ago
f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
9332b7d Reland "Update rtt on audio only calls". by michaelt · 8 years ago
78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
6287e82 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 8 years ago
9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
982bf89 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
e0729c5 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
e035e2d Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets. by terelius · 8 years ago
26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 8 years ago
a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
86cc6ff Variable audio bitrate. by mflodman · 8 years ago
9421853 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 8 years ago
1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 8 years ago
4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 8 years ago
8842c3e Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 8 years ago
3ecb5c8 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ ) by solenberg · 8 years ago
8886c81 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 8 years ago
fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 8 years ago
b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
85a0496 Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
c7a8b08 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago