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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
78ac89b82ff05aeeeb3a6de14f644abdcb20b9b9
/
audio
001546d
Break up rtc_event_log_api to solve circular dependencies.
by Qingsi Wang
· 7 years ago
f120cba
Delete AudioMonitor and related code.
by Niels Möller
· 7 years ago
65ce311
Removing useless dependencies on //testing/gmock.
by Mirko Bonadei
· 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
d8b041c
Ignore extra arguments in low_bandwidth_audio_test.
by Edward Lemur
· 7 years ago
649a385
Removes usage of analog AGC.
by henrika
· 7 years ago
90ea504
Delete Channel::OnRecoveredPacket.
by Niels Möller
· 7 years ago
98d4036
Make it possible to run low_bandwidth_audio_test on Android swarming.
by Edward Lemur
· 7 years ago
b401771
Store JSON perf results for low_bandwidth_audio_test.
by Edward Lemur
· 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
a7f2d84
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"""
by Per Kjellander
· 7 years ago
c73e1f4
Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
by Per Kjellander
· 7 years ago
588c548
GN rtc_* templates: Set default visibility to webrtc_root + "/*"
by Karl Wiberg
· 7 years ago
24722b3
Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator."
by Seth Hampson
· 7 years ago
731082c
Reland: Add mock_rtc_event_log.h.
by Patrik Höglund
· 7 years ago
5a25ab2
Revert "Add mock_rtc_event_log.h."
by Edward Lemur
· 7 years ago
63aea46
Add mock_rtc_event_log.h.
by Patrik Höglund
· 7 years ago
94dc177
Add mock_bitrate_controller.h.
by Patrik Höglund
· 7 years ago
6213929
Add missing files to audio_processing.
by Patrik Höglund
· 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
d524751
Replace VoEBase::[Start/Stop]Playout().
by Fredrik Solenberg
· 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
3e11343
Fix circular dependencies in webrtc_common.
by Patrik Höglund
· 7 years ago
a8005cf
Fix circular dependencies between optional, array_view, and rtc_base.
by Patrik Höglund
· 7 years ago
d37709b
Revert "Fix circular dependencies between optional, array_view, and rtc_base."
by Patrik Höglund
· 7 years ago
a9e0924
Fix circular dependencies between optional, array_view, and rtc_base.
by Patrik Höglund
· 7 years ago
cedd351
Do not add audio bitrate observer if TWCC sending is not supported
by Alex Narest
· 7 years ago
b5728d9
Stop using public_deps in modules/rtp_rtcp.
by Mirko Bonadei
· 7 years ago
5e849cf
Stop using public_deps in audio/utility.
by Mirko Bonadei
· 7 years ago
56d4609
Use the new AudioProcessing statistics everywhere.
by Ivo Creusen
· 7 years ago
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