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path: root/voice_engine/channel.cc
AgeCommit message (Expand)Author
2014-10-10Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. xians@webrtc.org
2014-10-09Estimating NTP time with a given RTT.minyue@webrtc.org
2014-10-01Reland "Remove DTMF status methods from Voice Engine" r7276henrik.lundin@webrtc.org
2014-09-23Revert "Remove DTMF status methods from Voice Engine" r7276henrik.lundin@webrtc.org
2014-09-23Remove DTMF status methods from Voice Enginehenrik.lundin@webrtc.org
2014-09-22Remove Get/SetNetEQPlayoutMode APIshenrik.lundin@webrtc.org
2014-09-11Calculating round-trip-time in send-only channel in VoE.minyue@webrtc.org
2014-09-03Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRateminyue@webrtc.org
2014-09-02Fix race in Voice Engine's Channel where it accesses RemoteNtpTimeEstimator f...stefan@webrtc.org
2014-08-12Adding SetOpusMaxBandwidth in VoE and ACMminyue@webrtc.org
2014-08-06Fixing two bugs in voe_cmd_test.minyue@webrtc.org
2014-07-25Remove timestamp retreival warning/error.turaj@webrtc.org
2014-07-16Raw packet loss rate reported by RTP_RTCP module may vary too drastically ove...minyue@webrtc.org
2014-07-11Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.tommi@webrtc.org
2014-07-11Remove the send-side cname getter APIs from voice and video engine.stefan@webrtc.org
2014-06-05Fix the chain that propagates the audio frame's rtp and ntp timestamp including:wu@webrtc.org
2014-06-05Have RTX be enabled by setting an RTX payload type instead of by setting an R...stefan@webrtc.org
2014-05-28This CL is to adding feedback of packet loss rate to encoder in voice engine....minyue@webrtc.org
2014-05-231. Make a clear distinction between codec internal FEC and RED, confusing men...minyue@webrtc.org
2014-05-20Calculate capture ntp timestamp in local timebase for decoded audio frame.wu@webrtc.org
2014-05-19Add interface to propagate audio capture timestamp to the renderer.wu@webrtc.org
2014-05-14Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.andrew@webrtc.org
2014-05-12Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation ...henrika@webrtc.org
2014-05-12Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operat...henrika@webrtc.org
2014-05-05Allow the RTP level indicator computation to work at any sample rate.andrew@webrtc.org
2014-04-25Replace scoped_array<T> with scoped_ptr<T[]>.andrew@webrtc.org
2014-04-24* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.wu@webrtc.org
2014-04-22Reland "Stop using ACM factory in VoiceEngine"henrik.lundin@webrtc.org
2014-04-22Revert "Stop using ACM factory in VoiceEngine"henrik.lundin@webrtc.org
2014-04-22Stop using ACM factory in VoiceEnginehenrik.lundin@webrtc.org
2014-04-17Removes parts of the VoEBase sub API as part of a clean-up operation where th...henrika@webrtc.org
2014-04-17Removes VoECodec sub API as part of a clean-up operation where the goal is to...henrika@webrtc.org
2014-04-16Re-enable AGC tests:aluebs@webrtc.org
2014-04-14Removes VoECallReport sub API as part of a clean-up operation where the goal ...henrika@webrtc.org
2014-04-08Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.andresp@webrtc.org
2014-04-03Consolidate audio conversion from Channel and TransmitMixer.andrew@webrtc.org
2014-03-31VoE Channel: Don't register codecs when stopping receiverhenrik.lundin@webrtc.org
2014-03-24VoE changes to allow forwarding of packets from VoE to ViE BWE.solenberg@webrtc.org
2014-03-20Prevent playout delay wrap-around in VoiceEnginehenrik.lundin@webrtc.org
2014-03-18Resolves TSan v2 warnings in voe_auto_test.henrika@webrtc.org
2014-03-11Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instea...braveyao@webrtc.org
2014-03-06Help to land 7969005 on behalf of solenberg. The review and try is done in 79...wu@webrtc.org
2014-02-19Removes VoERTP_RTCP::InsertExtraRTPPacket.henrika@webrtc.org
2014-02-18Remove external encryption API for VoE.solenberg@webrtc.org
2014-01-07Remove the requirement to call set_sample_rate_hz and friends.andrew@webrtc.org
2013-12-19Add callbacks for receive channel RTCP statistics.sprang@webrtc.org
2013-12-13Fix jitter buffer delay estimate.turaj@webrtc.org
2013-12-13Update talk to 58174641 together with http://review.webrtc.org/4319005/.wu@webrtc.org
2013-11-08Fix for making sure that the packet in order checks are done prior to updatin...stefan@webrtc.org
2013-10-22Upgrade scoped_ptr to Chromium's latest version.andrew@webrtc.org
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