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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
a21090b7706eb6577f4b305f3e00b04fe833c50d
/
audio
/
channel_proxy.h
bb50ce5
Wire up MID send value to the PeerConnection API
by Steve Anton
· 7 years ago
5f22365
Remove unnecessary proxy+lock code around RtcpRttStats pointer
by Tommi
· 7 years ago
f120cba
Delete AudioMonitor and related code.
by Niels Möller
· 7 years ago
a8b7c7f
Move remaining traces of VoiceEngine
by Fredrik Solenberg
· 7 years ago
[Renamed (97%) from voice_engine/channel_proxy.h]
90ea504
Delete Channel::OnRecoveredPacket.
by Niels Möller
· 7 years ago
8f5787a
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
by Fredrik Solenberg
· 7 years ago
d524751
Replace VoEBase::[Start/Stop]Playout().
by Fredrik Solenberg
· 7 years ago
aaedf75
Replace VoEBase::[Start/Stop]Send().
by Fredrik Solenberg
· 7 years ago
2a87797
Remove voe::TransmitMixer
by Fredrik Solenberg
· 7 years ago
1c239d4
Remove voe::Statistics.
by solenberg
· 7 years ago
2397b9a
Remove voe::OutputMixer and AudioConferenceMixer.
by solenberg
· 7 years ago
dd3abbb
Remove VoERTP_RTCP.
by solenberg
· 7 years ago
6dc2038
Remove VoECodec.
by solenberg
· 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/voice_engine/channel_proxy.h]
e1198e0
Add new ANA stats to the old GetStats() to count the number of actions taken by each controller.
by ivoc
· 7 years ago
e76bd3a
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
· 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
0f15f92
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
by nisse
· 7 years ago
eb1fde4
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
by ossu
· 8 years ago
20a4b3f
Injectable audio encoders: WebRtcVoiceEngine and company
by ossu
· 8 years ago
8d609f6
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
by hbos
· 8 years ago
fbcc5cb
Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
by olka
· 8 years ago
292084c
Added the GetSources() to the RtpReceiverInterface and implemented
by zhihuang
· 8 years ago
1ffbd6c
Injectable audio encoders: voice_engine/channel changes.
by ossu
· 8 years ago
fdbfdc9
Let PacketRouter separate send and receive modules.
by nisse
· 8 years ago
1c07c70
Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"
by kwiberg
· 8 years ago
b8f9a32
Define RtpTransportControllerSendInterface.
by nisse
· 8 years ago
670a7f3
Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ )
by kwiberg
· 8 years ago
1724cfb
WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
by kwiberg
· 8 years ago
dadb4dc
Allow ANA to receive RPLR (recoverable packet loss rate) indications
by elad.alon
· 8 years ago
d12a8e1
Attach TransportFeedbackPacketLossTracker to ANA (PLR only)
by elad.alon
· 8 years ago
8d73f8c
Remove VoEVolumeControl interface.
by solenberg
· 8 years ago
796b8f9
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
by solenberg
· 8 years ago
657bab2
Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.
by nisse
· 8 years ago
08b19df
Remove VoEVideoSync interface.
by solenberg
· 8 years ago
7de8d64
Wire up audio packet loss to BWE.
by stefan
· 8 years ago
bd9a77f
Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream.
by solenberg
· 8 years ago
3ebbcb5
Stop using VoEVideoSync in Call/VideoReceiveStream.
by solenberg
· 8 years ago
d32bf75
Pass SdpAudioFormat through Channel, without converting to CodecInst
by kwiberg
· 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
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
7602aab
Remove usage of VoEBase::AssociateSendChannel() from WVoMC.
by solenberg
· 8 years ago
79e0588
Set actual transport overhead in rtp_rtcp
by michaelt
· 8 years ago
b521aa7
Clean up abs-send-time for audio.
by stefan
· 8 years ago
6b825df
Using AudioOption to enable audio network adaptor.
by minyue
· 8 years ago
051f678
Add a NeededFrequency() method to the AudioMixer::Source interface.
by aleloi
· 8 years ago
6c27849
Move audio frame memory handling inside AudioMixer.
by aleloi
· 8 years ago
aed581a
Made AudioReceiveStream a mixer participant.
by aleloi
· 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
86cc6ff
Variable audio bitrate.
by mflodman
· 8 years ago
14d5dbe
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
· 8 years ago
9e03c3b
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
by ivoc
· 8 years ago
1895526
Move RtcEventLog object from inside VoiceEngine to Call.
by Ivo Creusen
· 8 years ago
217fb66
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
by solenberg
· 8 years ago
9421853
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
by solenberg
· 8 years ago
971cab0
Configure VoE NACK through AudioSendStream::Config, for send streams.
by solenberg
· 8 years ago
29b1a8d
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
by ossu
· 8 years ago
3d7db26
Switch voice transport to use Call and Stream instead of VoENetwork.
by mflodman
· 9 years ago
d53a3f9
Early initialize recording on the ADM from WebRtcVoiceMediaChannel.
by solenberg
· 9 years ago
ff97631
- Add temporary VoEBase::audio_device_module() method.
by solenberg
· 9 years ago
8842c3e
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
by solenberg
· 9 years ago
3ecb5c8
Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ )
by solenberg
· 9 years ago
8886c81
- Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.
by solenberg
· 9 years ago
b7f89d6
Replace scoped_ptr with unique_ptr in webrtc/voice_engine/
by kwiberg
· 9 years ago
bba9dec
Use separate rtp module lists for send and receive in PacketRouter.
by stefan
· 9 years ago
3313ec9
Enable transport seq num extension on receive channel to suppress log warning.
by stefan
· 9 years ago
2d110be
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
e591f93
Storing raw audio sink for default audio track.
by deadbeef
· 9 years ago
f888bb5
Support for unmixed remote audio into tracks.
by Tommi
· 9 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
358057b
Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream.
by 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