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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
d5bc865f136cc03d5c11ba8447728995f925cf69
/
audio
/
audio_send_stream.cc
fa4e185
Delete class voe::RtcEventLogProxy
by Niels Möller
· 6 years ago
848d6d3
Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver.
by Niels Möller
· 6 years ago
bbbe4e1
Better handle target audio bitrate allocation.
by Alex Narest
· 6 years ago
bcf9180
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
by Alex Narest
· 6 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 6 years ago
b9b146c
Replace rtc::Optional with absl::optional in audio, call and video
by Danil Chapovalov
· 6 years ago
867e510
Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled.
by Alex Narest
· 6 years ago
24ad720
Uses config struct with bitrate allocator.
by Sebastian Jansson
· 6 years ago
abbe841
This CL removes all usages of our custom ostream << overloads.
by Jonas Olsson
· 6 years ago
003930a
Fix MID not always getting set with audio
by Steve Anton
· 6 years ago
bb50ce5
Wire up MID send value to the PeerConnection API
by Steve Anton
· 6 years ago
5f22365
Remove unnecessary proxy+lock code around RtcpRttStats pointer
by Tommi
· 6 years ago
77490b9
Pass a real audio codec pair ID to encoders that we create
by Karl Wiberg
· 7 years ago
763e947
Reporting packet feedback availability in AudioSendStream
by Sebastian Jansson
· 7 years ago
98cd810
Production code: Pass codec ID argument to audio codecs
by Karl Wiberg
· 7 years ago
3c24ea8
Removed SetTransportOverhead in transport controller.
by Sebastian Jansson
· 7 years ago
4c1ffb8
Removing access to pacer in rtp controller.
by Sebastian Jansson
· 7 years ago
e4be6da
Removing access to send side cc in rtp controller.
by Sebastian Jansson
· 7 years ago
06953ba
Move AudioSendStream lifetime reporting into destructor
by Sam Zackrisson
· 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
8f5787a
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
by Fredrik Solenberg
· 7 years ago
24722b3
Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator."
by Seth Hampson
· 7 years ago
8b77aea
Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator."
by Lu Liu
· 7 years ago
d2b912a
Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator.
by Seth Hampson
· 7 years ago
f85e31b
Don't (re-)configure BitrateObserver unless already sending
by Oskar Sundbom
· 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
cedd351
Do not add audio bitrate observer if TWCC sending is not supported
by Alex Narest
· 7 years ago
56d4609
Use the new AudioProcessing statistics everywhere.
by Ivo Creusen
· 7 years ago
2707fb2
Optional: Use nullopt and implicit construction in /audio
by Oskar Sundbom
· 7 years ago
8d9c540
Deprecated BitrateController::CreateRtcpBandwidthObserver.
by Sebastian Jansson
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 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
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
b3944f0
Media track ID visibility at BWE level
by Alex Narest
· 7 years ago
90e1f53
Fix potentional race in AudioSendStream constructor
by Danil Chapovalov
· 7 years ago
1c239d4
Remove voe::Statistics.
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/audio/audio_send_stream.cc]
e1198e0
Add new ANA stats to the old GetStats() to count the number of actions taken by each controller.
by ivoc
· 7 years ago
3651fdd
Uncomment thread-check in AudioSendStream::OnPacketFeedbackVector()
by eladalon
· 7 years ago
8de1826
Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
by minyue-webrtc
· 7 years ago
7df370b
Revert "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
by Minyue Li
· 7 years ago
4a88120
Allow AudioSendStream to reconfig AudioNetworkAdaptor
by minyue-webrtc
· 7 years ago
abbc430
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
by eladalon
· 7 years ago
c58f8c0
Adds a histogram metric tracking for how long audio RTP packets are sent
by saza
· 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
1129df2
Always ResetSenderCongestionControlObjects before RegisterEtc...
by ossu
· 7 years ago
a9cc40b
Allow an external audio processing module to be used in WebRTC
by peah
· 7 years ago
edd6eea
Rename elad.alon to eladalon, to avoid confusion between repositories.
by eladalon
· 7 years ago
c3d4b48
Store/restore RTP state for audio streams with same SSRC within a call
by ossu
· 7 years ago
93e4522
Renaming probing_interval to bwe_period globally.
by minyue
· 7 years ago
48368ad
Fixing video loopback test with encoder factory.
by minyue
· 7 years ago
3b9ff38
Have AudioSendStream register CNG payload types with the RtpRtcpModule.
by ossu
· 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
fca900a
Fix two invalid DCHECKs related to audio BWE.
by stefan
· 7 years ago
fdbfdc9
Let PacketRouter separate send and receive modules.
by nisse
· 7 years ago
4e76451
Fix UT failure by temporarily uncommenting
by elad.alon
· 7 years ago
b8f9a32
Define RtpTransportControllerSendInterface.
by nisse
· 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
559af38
Split CongestionController into send- and receive-side classes.
by nisse
· 8 years ago
5bbf43f
Move delay_based_bwe_ into CongestionController
by elad.alon
· 8 years ago
796b8f9
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
by solenberg
· 8 years ago
06f240b
Clean out platform specific things from voice_engine_defines.h.
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
55d1ebb
Enable periodic bitrate probing when application limited for audio BWE.
by stefan
· 8 years ago
f4caaab
Fix for bwe with overhead on audio only calls.
by michaelt
· 8 years ago
4e477a1
Added a new echo likelihood stat that reports the maximum value from a previous time period.
by ivoc
· 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
0245da0
Move ownership of PacketRouter from CongestionController to Call.
by nisse
· 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
1acfbd2
Expose RtpCodecParameters to VoiceMediaInfo stats.
by hbos
· 8 years ago
d4adce4
Remove Absolute Send Time from list of supported header extensions for audio streams.
by solenberg
· 8 years ago
ffbbcac
Support multiple timestamp rates for sending DTMF.
by solenberg
· 8 years ago
7aba029
Make use of new APM statistics interface.
by ivoc
· 8 years ago
6f0b9fd
Allowing resetting of AudioNetworkAdaptor in AudioSendStream.
by minyue
· 8 years ago
79e0588
Set actual transport overhead in rtp_rtcp
by michaelt
· 8 years ago
10cbb46
Fixing config for Audio BWE.
by minyue
· 8 years ago
572ae12
Fix crash when registering abs-send-time to AudioSend/ReceiveStream.
by stefan
· 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
940b6d6
Clean up logging in AudioSendStream::SetupSendCodec().
by solenberg
· 8 years ago
189f9b1
Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ )
by terelius
· 8 years ago
1836fd6
Clean up logging in AudioSendStream::SetupSendCodec().
by solenberg
· 8 years ago
8c63a82
Add a placeholder stat for logging the estimated residual echo likelihood.
by ivoc
· 8 years ago
7a97344
Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.
by minyue
· 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
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