aboutsummaryrefslogtreecommitdiff
path: root/webrtc/voice_engine/channel.cc
AgeCommit message (Expand)Author
2016-01-13Revert of Storing raw audio sink for default audio track. (patchset #7 id:120...deadbeef
2016-01-13Storing raw audio sink for default audio track.deadbeef
2016-01-13Convert channel counts to size_t.Peter Kasting
2016-01-12Wire-up BWE feedback for audio receive streams.Stefan Holmer
2015-12-12Support for unmixed remote audio into tracks.Tommi
2015-12-10modules/rtp_rtcp/include folder cleared of lint warningsdanilchap
2015-12-07Prepare the AudioSendStream to be hooked up to send-side BWE.Stefan Holmer
2015-12-04- Remove calls to VoEDtmf from WVoE/MC.Fredrik Solenberg
2015-11-27Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream.solenberg
2015-11-27Use webrtc/base/logging.h for voice_engine.pbos
2015-11-04modules: more interface -> include renamesHenrik Kjellander
2015-11-03Let AudioCodingModule::SendCodec return Maybe<CodecInst>kwiberg
2015-11-01Delete a chain of methods in ViE, VoE and ACMhenrik.lundin
2015-10-28system_wrappers: rename interface -> includeHenrik Kjellander
2015-10-14Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.noahric
2015-10-02Add a PacketOptions struct to webrtc::Transport.stefan
2015-10-02Unify newapi::RtcpMode and RTCPMethod.pbos
2015-09-28Unify Transport and newapi::Transport interfaces.pbos
2015-09-23Fix the maximum native sample rate in AudioProcessingAlejandro Luebs
2015-09-23Reduce LS_INFO spam from voice_engine/.Peter Boström
2015-09-18ACM: Remove functions related to DTMFhenrik.lundin
2015-09-17Remove channel ids from various interfaces.Peter Boström
2015-09-17Added support for logging the SSRC corresponding to AudioPlayout events.Ivo Creusen
2015-09-17Remove the preprocessor symbol WEBRTC_CODEC_AVT (it was always defined)kwiberg
2015-09-09Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated t...ivoc
2015-08-31Stylizing AudioConferenceMixer.minyuel
2015-08-24Update a ton of audio code to use size_t more correctly and in general reducePeter Kasting
2015-08-13Adding locking to webrtc::voe::Channel to fix race conditionsdeadbeef
2015-07-07Remove ResetStatistics from RTP feedback.pbos
2015-06-11Match existing type usage better.Peter Kasting
2015-06-01Add options for NetEq fast accelerate mode through libjingleHenrik Lundin
2015-05-20Remove rtp_rtcp/ dump functionality.Peter Boström
2015-05-13Propagating RTT from send-only channel to receive-only channel.Minyue
2015-05-12Remove VideoEngine interfaces.Peter Boström
2015-05-11Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."Henrik Lundin
2015-05-11Setting OPUS_SIGNAL_VOICE when enable DTX.Minyue Li
2015-05-10Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."Henrik Lundin
2015-05-10Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."Henrik Lundin
2015-05-08Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."Henrik Lundin
2015-05-08Adding a new constraint to set NetEq buffer capacity from peerconnectionHenrik Lundin
2015-04-29Added SetBitRate function to VoE API to allow changing the audio bitrate.Ivo Creusen
2015-03-30Introduce CodecManager and move code from AudioCodingModuleImplHenrik Lundin
2015-03-13Supporting Opus DTX in Voice Engine.minyue@webrtc.org
2015-03-06Remove WebRtcACMEncodingTypehenrik.lundin@webrtc.org
2015-03-04Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macrokjellander@webrtc.org
2015-03-02Remove AudioCodingModule::Process()henrik.lundin@webrtc.org
2015-02-27ProcessThread improvements.tommi@webrtc.org
2015-02-26Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going awaykwiberg@webrtc.org
2015-02-23Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :pkasting@chromium.org
2015-02-18Adding two new stats to VoiceReceiverInfominyue@webrtc.org
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-25 22:02:51 +0000