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src
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webrtc
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4cd0b10627c8ac05bfeacc636e4c5d8869b2a848
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audio
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audio_receive_stream.cc
443f9a9
Wiring discard rate of audio FEC/RED packets up to StatsReport.
by minyue-webrtc
· 7 years ago
94ac82f
Add stats totalSamplesReceived and concealedSamples
by Steve Anton
· 7 years ago
ea04cc0
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
by eladalon
· 7 years ago
d1701d0
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
by zstein
· 8 years ago
76de83e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 8 years ago
bc32410
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 8 years ago
60154fd
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 8 years ago
04ef6b0
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
by nisse
· 8 years ago
69579d7
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
by hbos
· 8 years ago
fee642e
Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
by olka
· 8 years ago
b5eeeee
Added the GetSources() to the RtpReceiverInterface and implemented
by zhihuang
· 8 years ago
5e9233b
Let PacketRouter separate send and receive modules.
by nisse
· 8 years ago
afd1255
Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"
by kwiberg
· 8 years ago
52b9df6
Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ )
by kwiberg
· 8 years ago
135be28
WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
by kwiberg
· 8 years ago
bf77153
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
by solenberg
· 8 years ago
9617a87
Replace NULL with nullptr or null in webrtc/audio/ and common_audio/.
by deadbeef
· 8 years ago
95a13b4
Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.
by nisse
· 8 years ago
ac16f1b
Remove VoEVideoSync interface.
by solenberg
· 8 years ago
49cdd35
Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.
by nisse
· 8 years ago
f65fb4c
Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/
by solenberg
· 8 years ago
6045993
Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream.
by solenberg
· 8 years ago
a0add70
Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ )
by nisse
· 8 years ago
5cacd3d
Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ )
by nisse
· 8 years ago
40ab430
Always call RemoteBitrateEstimator::IncomingPacket from Call.
by nisse
· 8 years ago
9551ae0
Stop using VoEVideoSync in Call/VideoReceiveStream.
by solenberg
· 8 years ago
bc2b639
Pass SdpAudioFormat through Channel, without converting to CodecInst
by kwiberg
· 8 years ago
800a8f1
Support external audio mixer in webrtc 2.
by gyzhou
· 8 years ago
818a3ee
Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ )
by gyzhou
· 8 years ago
0808545
Support external audio mixer in webrtc.
by gyzhou
· 8 years ago
61e42da
Move ownership of PacketRouter from CongestionController to Call.
by nisse
· 8 years ago
441ecb3
Replace AudioConferenceMixer with AudioMixer.
by aleloi
· 8 years ago
9a1d49f
Expose RtpCodecParameters to VoiceMediaInfo stats.
by hbos
· 8 years ago
ac14ca2
Remove Absolute Send Time from list of supported header extensions for audio streams.
by solenberg
· 8 years ago
dbe2c77
Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
by solenberg
· 8 years ago
def0180
Fix crash when registering abs-send-time to AudioSend/ReceiveStream.
by stefan
· 8 years ago
e9523f0
Clean up abs-send-time for audio.
by stefan
· 8 years ago
b87bc5d
Add a NeededFrequency() method to the AudioMixer::Source interface.
by aleloi
· 8 years ago
fbd0246
Move audio frame memory handling inside AudioMixer.
by aleloi
· 8 years ago
b4ac6b0
Made AudioReceiveStream a mixer participant.
by aleloi
· 8 years ago
d0dcb3b
Restarting channel when swapping AudioReceiveStreams in WebrtcVoE.
by aleloi
· 8 years ago
4450c27
Add new decoding statistics for muted output
by henrik.lundin
· 8 years ago
cf82062
Move webrtc/audio_*.h to webrtc/api/call
by kjellander
· 8 years ago
762f0f9
Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.
by aleloi
· 8 years ago
23ea12e
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
by ivoc
· 9 years ago
822f09e
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
by ivoc
· 9 years ago
1e2f1e5
Move RtcEventLog object from inside VoiceEngine to Call.
by Ivo Creusen
· 9 years ago
6ba7dfc
Reland of move audio/video distinction for probe packets. (patchset #1 id:1 of https://codereview.webrtc.org/2086633002/ )
by pbos
· 9 years ago
f4d4351
Revert of Remove audio/video distinction for probe packets. (patchset #2 id:20001 of https://codereview.webrtc.org/2061193002/ )
by honghaiz
· 9 years ago
b0d0745
Remove audio/video distinction for probe packets.
by Peter Boström
· 9 years ago
784336a
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
by solenberg
· 9 years ago
ff1d51a
Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test.
by solenberg
· 9 years ago
e840777
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
by ossu
· 9 years ago
c4921f4
Remove use of RtpHeaderExtension and clean up
by isheriff
· 9 years ago
7738985
Delete all use of tick_util.h.
by Niels Möller
· 9 years ago
47a40a3
Remove webrtc/stream.h and unutilized inheritance.
by pbos
· 9 years ago
823f908
Switch voice transport to use Call and Stream instead of VoENetwork.
by mflodman
· 9 years ago
caa8176
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
by kjellander@webrtc.org
· 9 years ago
52cf08c
Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
by kjellander
· 9 years ago
0bb951e
Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
by kjellander@webrtc.org
· 9 years ago
abb2e3e
Replace scoped_ptr with unique_ptr in webrtc/audio/
by kwiberg
· 9 years ago
ead3cf2
Move congestion controller to a separate module.
by Stefan Holmer
· 9 years ago
484e0cd
Replace scoped_ptr with unique_ptr in webrtc/voice_engine/
by kwiberg
· 9 years ago
74c29e2
Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
by stefan
· 9 years ago
3dad57b
Use separate rtp module lists for send and receive in PacketRouter.
by stefan
· 9 years ago
9b91023
Enable transport seq num extension on receive channel to suppress log warning.
by stefan
· 9 years ago
e3f40fb
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
by deadbeef
· 9 years ago
cf354ef
Storing raw audio sink for default audio track.
by deadbeef
· 9 years ago
87f3db7
Wire-up BWE feedback for audio receive streams.
by Stefan Holmer
· 9 years ago
387e90b
Support for unmixed remote audio into tracks.
by Tommi
· 9 years ago
21ca0a4
Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream.
by solenberg
· 9 years ago
5bbf7f9
Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID.
by solenberg
· 9 years ago
775e132
Move some receive stream configuration into webrtc::AudioReceiveStream.
by solenberg
· 9 years ago
03d4810
Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail.
by solenberg
· 9 years ago
b0f22c5
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
by solenberg
· 9 years ago
78f65d0
system_wrappers: rename interface -> include
by Henrik Kjellander
· 9 years ago
1c28f5c
Implement AudioSendStream::GetStats().
by solenberg
· 9 years ago
10762d3
Re-Land: Implement AudioReceiveStream::GetStats().
by Fredrik Solenberg
· 9 years ago
0e9f679
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
by solenberg
· 9 years ago
bbb922f
Implement AudioReceiveStream::GetStats().
by Fredrik Solenberg
· 9 years ago
f863304
Log Call {audio, video} stream deletions.
by pbos
· 9 years ago
bf9f73c
Split webrtc/video into webrtc/{audio,call,video}.
by Peter Boström
· 9 years ago
[Renamed (98%) from video/audio_receive_stream.cc]
4f5e451
Fix BWE bug where audio has timestamps in us.
by Stefan Holmer
· 9 years ago
5c075c8
Add RTC_ prefix to (D)CHECKs and related macros.
by henrikg
· 9 years ago
30bf778
Wire up PacketTime to ReceiveStreams.
by stefan
· 9 years ago
374a570
Add support for transport wide sequence numbers
by sprang
· 9 years ago
95b9fbb
Control combined_audio_video_bwe with config bool.
by pbos
· 10 years ago
2fb88e4
Define Stream base classes
by Jelena Marusic
· 10 years ago
495b350
Base A/V synchronization on sync_labels.
by pbos
· 10 years ago
9dca5d7
Only use paced packets for estimating bitrate probes.
by Stefan Holmer
· 10 years ago
ee9b72e
VoE2 API draft
by Fredrik Solenberg
· 10 years ago
ad86786
Add AudioReceiveStream to Call API.
by Fredrik Solenberg
· 10 years ago