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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
29e7bee3308db6de1dfc113644a12661fa6ccdd7
/
audio
/
audio_send_stream_unittest.cc
fe617a3
Adding has_packet_feedback to LimitObserver callback.
by Sebastian Jansson
· 7 years ago
d6fbf2a
Tests: Pass codec ID argument to audio codecs
by Karl Wiberg
· 7 years ago
f69e768
Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 1.
by philipel
· 7 years ago
ef9daee
Using mock transport controller in audio unit tests.
by Sebastian Jansson
· 7 years ago
41f16be
Silencing warnings in audio send stream unit tests.
by Sebastian Jansson
· 7 years ago
97f61ea
Moved bitrate configuration to rtp controller
by Sebastian Jansson
· 7 years ago
1896cec
Removed dependencies from audio send stream unit test
by Sebastian Jansson
· 7 years ago
06953ba
Move AudioSendStream lifetime reporting into destructor
by Sam Zackrisson
· 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
f85e31b
Don't (re-)configure BitrateObserver unless already sending
by Oskar Sundbom
· 7 years ago
2a87797
Remove voe::TransmitMixer
by Fredrik Solenberg
· 7 years ago
56d4609
Use the new AudioProcessing statistics everywhere.
by Ivo Creusen
· 7 years ago
d319534
Move ADM initialization into WebRtcVoiceEngine
by Fredrik Solenberg
· 7 years ago
2707fb2
Optional: Use nullopt and implicit construction in /audio
by Oskar Sundbom
· 7 years ago
1c239d4
Remove voe::Statistics.
by solenberg
· 7 years ago
fc3a2e3
Remove the VoiceEngineObserver callback interface.
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_unittest.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
5c8942a
Move PacedSender ownership to RtpTransportControllerSend.
by Stefan Holmer
· 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
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
c3d4b48
Store/restore RTP state for audio streams with same SSRC within a call
by ossu
· 7 years ago
8c96a14
Simple tests for Call::SetBitrateConfig.
by zstein
· 7 years ago
7cb69d5
This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008).
by zstein
· 7 years ago
eb1fde4
Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
by ossu
· 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
fdbfdc9
Let PacketRouter separate send and receive modules.
by nisse
· 8 years ago
b8f9a32
Define RtpTransportControllerSendInterface.
by nisse
· 8 years ago
40854ea
Reland of Delete class MockCongestionController. (patchset #1 id:1 of https://codereview.webrtc.org/2762133003/ )
by nisse
· 8 years ago
e27f1e7
Revert of Delete class MockCongestionController. (patchset #4 id:60001 of https://codereview.webrtc.org/2762023004/ )
by skvlad
· 8 years ago
d19bcb7
Delete class MockCongestionController.
by nisse
· 8 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
fb1fa44
Remove MockRemoteBitrateObserver (unused)
by elad.alon
· 8 years ago
796b8f9
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
by solenberg
· 8 years ago
922246a
Replace NULL with nullptr or null in webrtc/audio/ and common_audio/.
by deadbeef
· 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
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
10111bc
Passed AudioMixer to AudioState::Config.
by aleloi
· 8 years ago
dd31071
Added an empty AudioTransportProxy to AudioState.
by aleloi
· 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
10cbb46
Fixing config for Audio BWE.
by minyue
· 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
2d81eb3
Fix BWE simulations so that it uses the delay based BWE.
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
ac9f876
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
by kwiberg
· 8 years ago
77eab70
Enable the -Wundef warning for clang
by kwiberg
· 8 years ago
e035e2d
Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets.
by terelius
· 8 years ago
26091b1
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
by perkj
· 8 years ago
8eb37a3
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
by perkj
· 8 years ago
cc16836
- Add task queue to Call with the intent of replacing the use of one of the process threads.
by perkj
· 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
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
6806136
Remove RED support from WebRtcVoiceEngine/MediaChannel
by kwiberg
· 8 years ago
6f8d686
Remove use of RtpHeaderExtension and clean up
by isheriff
· 8 years ago
ec81bcd
Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate
by perkj
· 8 years ago
e30c272
Revert "Reland of Remove SendPacer from ViEEncoder
by perkj
· 8 years ago
28a4456
Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )"
by Per
· 8 years ago
825eb58
Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )
by perkj
· 8 years ago
857c5cc
Remove SendPacer from ViEEncoder
by perkj
· 8 years ago
3d7db26
Switch voice transport to use Call and Stream instead of VoENetwork.
by mflodman
· 8 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
80e1207
Move congestion controller to a separate module.
by Stefan Holmer
· 9 years ago
789ba92
Simplify CongestionController.
by Stefan Holmer
· 9 years ago
58c664c
Clean up of CongestionController.
by Stefan Holmer
· 9 years ago
bba9dec
Use separate rtp module lists for send and receive in PacketRouter.
by stefan
· 9 years ago
3842c5c
Wire-up BWE feedback for audio receive streams.
by Stefan Holmer
· 9 years ago
25702cb
Misc. small cleanups.
by pkasting
· 9 years ago
7623ce4
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
by Peter Boström
· 9 years ago
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