Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
2870b0a57e68be61331ba9b8b38e762d41a242ba
/
audio
/
channel.cc
f120cba
Delete AudioMonitor and related code.
by Niels Möller
· 7 years ago
24ea822
Remove logging in audio/* from release builds.
by Jonas Olsson
· 7 years ago
a8b7c7f
Move remaining traces of VoiceEngine
by Fredrik Solenberg
· 7 years ago
[Renamed (98%) from voice_engine/channel.cc]
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
3b903d0
Reconfigure, not reconstruct, AudioReceiveStreams.
by Fredrik Solenberg
· 7 years ago
2a87797
Remove voe::TransmitMixer
by Fredrik Solenberg
· 7 years ago
abbff89
Add new UMA metric for NetEq target buffer delay
by Henrik Lundin
· 7 years ago
606c882
Optional: Use nullopt and implicit construction in /voice_engine
by Oskar Sundbom
· 7 years ago
55900fd
Move APM initialization into WebRtcVoiceEngine
by Fredrik Solenberg
· 7 years ago
de93943
Revert "Revert "Encode log events periodically instead of for every event.""
by Bjorn Terelius
· 7 years ago
33c5c7f
Revert "Encode log events periodically instead of for every event."
by Zhi Huang
· 7 years ago
b154c27
Encode log events periodically instead of for every event.
by Bjorn Terelius
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 7 years ago
9bde6b7
Add new UMA metric for the audio receiver delay
by Henrik Lundin
· 7 years ago
78609d5
Reland of BWE allocation strategy
by Alex Narest
· 7 years ago
dc9ca93
Revert "BWE allocation strategy"
by Alex Narest
· 7 years ago
8818237
voe::Channel: Don't use CodecManager and RentACodec
by Karl Wiberg
· 7 years ago
a5fbc23
BWE allocation strategy
by Alex Narest
· 7 years ago
39260c4
Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."
by Lu Liu
· 7 years ago
54d1da1
BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
by Alex Narest
· 7 years ago
604c14d
Reland "Remove deprecated functions from RtcEventLog"
by Elad Alon
· 7 years ago
d312a91
Revert "Remove deprecated functions from RtcEventLog"
by Danil Chapovalov
· 7 years ago
5fd6e5e
Remove deprecated functions from RtcEventLog
by Elad Alon
· 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
83ccca1
Create and use RtcEventLogOutput for output
by Elad Alon
· 7 years ago
4a87e1c
Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead
by Elad Alon
· 7 years ago
ecc51e9
Change to LOG(...) logging in most of voice_engine/channel.cc
by Sam Zackrisson
· 7 years ago
440216f
Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets.
by Bjorn Terelius
· 7 years ago
1c239d4
Remove voe::Statistics.
by solenberg
· 7 years ago
c7b4a45
Remove various IDs:
by solenberg
· 7 years ago
4580217
Adds WebRTC.Audio.EncodingTaskQueueLatencyMs
by henrika
· 7 years ago
e423a9de
Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ )
by solenberg
· 7 years ago
2d0f775
Remove various IDs:
by solenberg
· 7 years ago
6df16bf
Remove unnecessary send codec initialization from voe::Channel.
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
73b60b8
Remove the redundant method GetPayloadSpecifics
by Karl Wiberg
· 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
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.cc]
479d3d7
Drop return value from RtpRtcp::IncomingRtcpPacket.
by nisse
· 7 years ago
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
529662a
Move array_view.h to webrtc/api/
by kwiberg
· 7 years ago
1acbd68
Move RtpExtension to api/ directory and config.h/.cc to call/.
by Stefan Holmer
· 7 years ago
da194e7
Delete remnants of RTX support in voice_engine.
by nisse
· 7 years ago
e9ef907
Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ )
by maxmorin
· 7 years ago
c251cb1
Add logging host lookups made by TurnPort to the RtcEventLog.
by jonaso
· 7 years ago
ee89e78
Replace CHECK(x && y) with two separate CHECK() calls
by kwiberg
· 7 years ago
3e69e5c
Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock
by srte
· 7 years ago
186d9c3
Renamed fields in common_types.h/RtcpStatistics.
by srte
· 7 years ago
822ff2b
Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates
by eladalon
· 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
950c1c9
TransmitMixer: Check GetSendCodec return value.
by ossu
· 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
36b1a5f
Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:
by yujo
· 7 years ago
76d29f9
Fix Channel::GetSendCodec when used together with SetEncoder.
by ossu
· 7 years ago
77cd58e
This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.
by perkj
· 7 years ago
f472699
Replace AudioSendStream::Config with rtclog::StreamConfig.
by perkj
· 7 years ago
ac8f52d
Replace AudioReceiveStream::Config with rtclog::StreamConfig.
by perkj
· 7 years ago
c0876aa
Replace VideoSendStream::Config with new rtclog::StreamConfig in RtcEventLog.
by perkj
· 7 years ago
09e71da
Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog.
by perkj
· 7 years ago
4515fa0
Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend()
by henrika
· 7 years ago
20a4b3f
Injectable audio encoders: WebRtcVoiceEngine and company
by ossu
· 7 years ago
cae45d0
Move RtpTransportControllerSend to a new file.
by nisse
· 7 years ago
92aef17
Replace Clock with timeutils in AudioEncoder.
by michaelt
· 7 years ago
1ffbd6c
Injectable audio encoders: voice_engine/channel changes.
by ossu
· 7 years ago
cde46b7
Resolve cyclic dependency between audio network adaptor and event log api
by michaelt
· 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
2877048
Experiment-driven configuration of PLR/RPLR-based FecController
by elad.alon
· 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
fe7dd6d
Remove VoEAudioProcessing interface.
by solenberg
· 8 years ago
67fdb80
Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ )
by oprypin
· 8 years ago
e47c1d3
Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ )
by oprypin
· 8 years ago
aebe55c
Enable cpplint and fix cpplint errors in webrtc/*audio
by oprypin
· 8 years ago
8d73f8c
Remove VoEVolumeControl interface.
by solenberg
· 8 years ago
f949000
Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo.
by elad.alon
· 8 years ago
dea489f
Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule.
by tommi
· 8 years ago
796b8f9
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
by solenberg
· 8 years ago
09f090c
Remove workaround for bug 6986
by kwiberg
· 8 years ago
3fd31fe
Fix TSAN race in webrtc::voe::Channel.
by hbos
· 8 years ago
32d0010
Add probe logging to RtcEventLog.
by philipel
· 8 years ago
8aadd50
Propagate packet pacing information to SendTimeHistory.
by philipel
· 8 years ago
657bab2
Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.
by nisse
· 8 years ago
7d59f6b
Reland of Delete class SSRCDatabase, and its global ssrc registry. (patchset #1 id:1 of https://codereview.webrtc.org/2700413002/ )
by nisse
· 8 years ago
Next »