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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
aea1d1ad3fa1d4225f581e39ceb040fa641b5c7f
/
audio
e40468b
Move some numeric utility code from rtc_base/ to rtc_base/numerics/
by Karl Wiberg
· 7 years ago
d319534
Move ADM initialization into WebRtcVoiceEngine
by Fredrik Solenberg
· 7 years ago
63e6072
Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine.
by Fredrik Solenberg
· 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
c0e6804
Fix deps of audio:audio_tests.
by Patrik Höglund
· 7 years ago
61a7b14
Removing conditional visibility.
by Mirko Bonadei
· 7 years ago
6d85252
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection AP (follow-up)
by henrika
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 7 years ago
5f6bf24
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)
by henrika
· 7 years ago
990d6b8
Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API"
by Mirko Bonadei
· 7 years ago
e4be4b7
Revert "Remove const from ThreadChecker in NullAudioPoller."
by Mirko Bonadei
· 7 years ago
54e41dd
Remove const from ThreadChecker in NullAudioPoller.
by Bjorn Terelius
· 7 years ago
90bace0
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
by henrika
· 7 years ago
9155e49
New classes RefCounter and RefCountedBase.
by Niels Möller
· 7 years ago
78609d5
Reland of BWE allocation strategy
by Alex Narest
· 7 years ago
6f72f56
Change return types of refcount methods.
by Niels Möller
· 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
245660a
Fix Gn untracked headers in webrtc/call.
by Mirko Bonadei
· 7 years ago
88b23f6
Fix flag name in low_bandwidth_audio_test.py
by Edward Lemur
· 7 years ago
7e3b569
Ignore swarming arguments in low_bandwidth_audio_test.py
by Edward Lemur
· 7 years ago
b0250f0
Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
by Edward Lemur
· 7 years ago
90e1f53
Fix potentional race in AudioSendStream constructor
by Danil Chapovalov
· 7 years ago
c3fa8e1
New method RtpReceiver::GetLatestTimestamps.
by Niels Möller
· 7 years ago
45a0b36
Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
by Edward Lemur
· 7 years ago
f4898a6
Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
by Edward Lemur
· 7 years ago
bb1222f
Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script."
by Edward Lemur
· 7 years ago
2019698
Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script.
by Edward Lemur
· 7 years ago
2011075
MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py.
by Edward Lemur
· 7 years ago
b0a0207
Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
by Gustaf Ullberg
· 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
4652e86
Disable flaky AudioStats.NoLoss test.
by solenberg
· 7 years ago
9a2e906
Added RTCMediaStreamTrackStats.concealmentEvents
by Gustaf Ullberg
· 7 years ago
18f5427
Remove voe_auto_test and add new tests to cover the missing cases.
by solenberg
· 7 years ago
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 7 years ago
5a6aa4f
Fix path to root in low_bandwidth_audio_test.py
by Henrik Kjellander
· 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