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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
5ff64831f9403357b39552702be641faaa73057f
/
audio
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