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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
53e048d83aa47de7bf98d1f10c667bd0e79551e7
/
audio
/
channel.h
a8b7c7f
Move remaining traces of VoiceEngine
by Fredrik Solenberg
· 7 years ago
[Renamed (98%) from voice_engine/channel.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
2a87797
Remove voe::TransmitMixer
by Fredrik Solenberg
· 7 years ago
55900fd
Move APM initialization into WebRtcVoiceEngine
by Fredrik Solenberg
· 7 years ago
8818237
voe::Channel: Don't use CodecManager and RentACodec
by Karl Wiberg
· 7 years ago
22ec952
Delete in_order argument to RtpReceiver::IncomingRtpPacket
by Niels Möller
· 7 years ago
c62f6c7
RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs
by Karl Wiberg
· 7 years ago
1c239d4
Remove voe::Statistics.
by solenberg
· 7 years ago
fc3a2e3
Remove the VoiceEngineObserver callback interface.
by solenberg
· 7 years ago
2397b9a
Remove voe::OutputMixer and AudioConferenceMixer.
by solenberg
· 7 years ago
946d886
Remove VoENetwork
by solenberg
· 7 years ago
dd3abbb
Remove VoERTP_RTCP.
by solenberg
· 7 years ago
6dc2038
Remove VoECodec.
by solenberg
· 7 years ago
b63310a
Remove VoEFile and things it uses.
by solenberg
· 7 years ago
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 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.h]
a37de39
Update thread annotiation macros to use RTC_ prefix
by danilchap
· 7 years ago
e1198e0
Add new ANA stats to the old GetStats() to count the number of actions taken by each controller.
by ivoc
· 7 years ago
84f6a3f
Move optional.h to webrtc/api/
by kwiberg
· 7 years ago
da194e7
Delete remnants of RTX support in voice_engine.
by nisse
· 7 years ago
3c45186
Move total audio energy and duration tracking to AudioLevel and protect with existing critial section.
by zstein
· 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
76d29f9
Fix Channel::GetSendCodec when used together with SetEncoder.
by ossu
· 7 years ago
30e8931
Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead.
by nisse
· 7 years ago
4515fa0
Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend()
by henrika
· 7 years ago
eb1fde4
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
by ossu
· 7 years ago
20a4b3f
Injectable audio encoders: WebRtcVoiceEngine and company
by ossu
· 7 years ago
8d609f6
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
by hbos
· 7 years ago
fbcc5cb
Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
by olka
· 7 years ago
292084c
Added the GetSources() to the RtpReceiverInterface and implemented
by zhihuang
· 7 years ago
1ffbd6c
Injectable audio encoders: voice_engine/channel changes.
by ossu
· 7 years ago
fdbfdc9
Let PacketRouter separate send and receive modules.
by nisse
· 7 years ago
ec6fbd2
Moves channel-dependent audio input processing to separate encoder task queue.
by henrika
· 7 years ago
1c07c70
Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"
by kwiberg
· 7 years ago
b8f9a32
Define RtpTransportControllerSendInterface.
by nisse
· 7 years ago
670a7f3
Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ )
by kwiberg
· 7 years ago
1724cfb
WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
by kwiberg
· 7 years ago
dadb4dc
Allow ANA to receive RPLR (recoverable packet loss rate) indications
by elad.alon
· 7 years ago
d12a8e1
Attach TransportFeedbackPacketLossTracker to ANA (PLR only)
by elad.alon
· 7 years ago
0a2391f
Add thread check to ModuleProcessThread::DeRegisterModule and remove all unnecessary locking that was there due to one implementation calling from a different thread.
by tommi
· 7 years ago
c6192a9
Remove VoENetEqStats interface.
by solenberg
· 7 years ago
fe7dd6d
Remove VoEAudioProcessing interface.
by solenberg
· 7 years ago
8d73f8c
Remove VoEVolumeControl interface.
by solenberg
· 7 years ago
92a7a18
Update formatting of AudioLevel class
by henrik.lundin
· 7 years ago
796b8f9
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
by solenberg
· 7 years ago
3fd31fe
Fix TSAN race in webrtc::voe::Channel.
by hbos
· 7 years ago
657bab2
Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.
by nisse
· 7 years ago
08b19df
Remove VoEVideoSync interface.
by solenberg
· 7 years ago
e374e01
Remove VoEExternalMedia interface.
by solenberg
· 7 years ago
81d93f3
Remove the unused and untested functions from VoERTP_RTCP.
by solenberg
· 7 years ago
7de8d64
Wire up audio packet loss to BWE.
by stefan
· 7 years ago
d32bf75
Pass SdpAudioFormat through Channel, without converting to CodecInst
by kwiberg
· 8 years ago
566d820
Update smoothed bitrate.
by michaelt
· 8 years ago
284542b
Make OverheadObserver::OnOverheadChanged count RTP headers only
by nisse
· 8 years ago
9774447
Move FilePlayer and FileRecorder to Voice Engine
by kwiberg
· 8 years ago
bf65be5
Wire-up audio BWE with overhead.
by michaelt
· 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
5049942
Refactor RMSLevel and give it new functionality
by henrik.lundin
· 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
2fedf9c
Smooth BWE and pass it to Audio Network Adaptor.
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
e566ac7
Remove voe::Channel::StopReceive() and associated logic.
by solenberg
· 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
e280cde
Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.
by ossu
· 8 years ago
7e30432
Hooking up audio network adaptor to VoE.
by minyue
· 8 years ago
e0729c5
Add RtcpRttStats to AudioStream
by michaelt
· 8 years ago
799a9d0
Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
by danilchap
· 8 years ago
2beb429
Remove unnecessary interface TelephoneEventHandler.
by solenberg
· 8 years ago
11ace15
The VoE functionality to apply receive-side processing to VoE channels is unused. I'm removing it so we can avoid instantiating a full APM per channel (and thus also for webrtc::AudioSendStream and webrtc::AudioReceiveStream), and then never use it.
by solenberg
· 8 years ago
88499ec
Moving/renaming webrtc/common.h.
by solenberg
· 8 years ago
b3e3001
Remove Channel::UpdatePacketDelay and some member variables
by henrik.lundin
· 8 years ago
a69d973
Move webrtc/audio_*.h to webrtc/api/call
by kjellander
· 8 years ago
e1f5b4a
voice_engine: Removed old variants of Channel constructor and CreateChannel
by ossu
· 8 years ago
5a25d95
FileRecorder + FilePlayer: Let Create functions return unique_ptr
by kwiberg
· 8 years ago
9d7eb13
Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ )
by kwiberg
· 8 years ago
427ce3d
Move FilePlayer and FileRecorder to Voice Engine
by kwiberg
· 8 years ago
c8c71f4
Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #6 id:100001 of https://codereview.webrtc.org/2240163002/ )
by kwiberg
· 8 years ago
dc65ea2
Move FilePlayer and FileRecorder to Voice Engine
by kwiberg
· 8 years ago
737336d
Add NACK rate throttling for audio channels.
by Erik Språng
· 8 years ago
85228d6
Regression test for issue where Opus DTX status was being forgotten.
by ivoc
· 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
e7edea9
Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #5 id:80001 of https://codereview.chromium.org/2037623002/ )
by kwiberg
· 8 years ago
65874b1
Move FilePlayer and FileRecorder to Voice Engine
by Karl Wiberg
· 8 years ago
9a38cab
Voice Engine: Remove RED support
by kwiberg
· 8 years ago
29b1a8d
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
by ossu
· 8 years ago
5f7cfa5
Moved CreateBuiltinDecoderFactory out to VoEBaseImpl.
by ossu
· 8 years ago
42dda50
Propagate muted info from VoE Channel to AudioConferenceMixer
by henrik.lundin
· 8 years ago
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