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path: root/talk/media/webrtc/fakewebrtcvoiceengine.h
AgeCommit message (Expand)Author
2016-01-13Convert channel counts to size_t.Peter Kasting
2016-01-12Remove additional channel constraints when Beamforming is enabled in AudioPro...aluebs
2015-12-19Revert of Reland "Added option to specify a maximum file size when recording ...ivoc
2015-12-19Reland "Added option to specify a maximum file size when recording an AEC dum...ivoc
2015-12-18Revert of Added option to specify a maximum file size when recording an AEC d...ivoc
2015-12-18Added option to specify a maximum file size when recording an AEC dump.ivoc
2015-12-16Bugfix that fixes the error where the audio processing module is calledpeah
2015-12-08Refactor handling of AudioOptions.solenberg
2015-12-04- Remove calls to VoEDtmf from WVoE/MC.Fredrik Solenberg
2015-12-02- Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused.solenberg
2015-11-27Using Rent-A-Codec for static Codec access in WVoE/MC.solenberg
2015-11-20Move some receive stream configuration into webrtc::AudioReceiveStream.solenberg
2015-11-16Move some send stream configuration into webrtc::AudioSendStream.solenberg
2015-11-03Changed FakeVoiceEngine into a MockVoiceEngine.Fredrik Solenberg
2015-10-27Implement AudioSendStream::GetStats().solenberg
2015-10-22Re-Land: Implement AudioReceiveStream::GetStats().Fredrik Solenberg
2015-10-20Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 o...solenberg
2015-10-20Implement AudioReceiveStream::GetStats().Fredrik Solenberg
2015-10-09Change SetOutputScaling to set a single level, not left/right levels.solenberg
2015-10-09Remove default receive channel from WVoE; baby step 1.solenberg
2015-10-07Use suffixed {uint,int}{8,16,32,64}_t types.Peter Boström
2015-10-02Removed unused API functions in AudioProcessing and AudioProcessingModulepeah
2015-09-23Improving support for Android Audio Effects in WebRTC.henrika
2015-09-23Remove the [Un]RegisterVoiceProcessor() API.Fredrik Solenberg
2015-09-23Remove VoEFile from VoeWrapper and the remaining places in libjingle where it...Fredrik Solenberg
2015-09-22Remove the SetLocalMonitor() API.solenberg
2015-09-17Add RTC_ prefix to (D)CHECKs and related macros.henrikg
2015-09-09Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated t...ivoc
2015-08-24Update a ton of audio code to use size_t more correctly and in general reducePeter Kasting
2015-08-14Integrate Intelligibility with APMekmeyerson
2015-07-23Allow more than 2 input channels in AudioProcessing.Michael Graczyk
2015-07-23Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 ...magjed
2015-07-23Allow more than 2 input channels in AudioProcessing.Michael Graczyk
2015-07-07audio_processing: Adds two UMA histograms logging delay jumps in AECBjorn Volcker
2015-06-01Add options for NetEq fast accelerate mode through libjingleHenrik Lundin
2015-05-26VoE: Remove unused interfacesJelena Marusic
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
2015-05-10Revert 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
2015-05-07Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/.Fredrik Solenberg
2015-05-07Remove unused API on WebRtcVoiceEngine.Fredrik Solenberg
2015-04-29Added SetBitRate function to VoE API to allow changing the audio bitrate.Ivo Creusen
2015-03-27Adding "usedtx" as Opus codec parameter.Minyue Li
2015-03-13Supporting Opus DTX in Voice Engine.minyue@webrtc.org
2015-03-04Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macrokjellander@webrtc.org
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