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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
5f6bf245066ccceecf208ecfefb7a603b15658f7
/
call
5f6bf24
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)
by henrika
· 7 years ago
de69145
Remove pbos@webrtc.org from all OWNERS.
by Peter Boström
· 7 years ago
990d6b8
Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API"
by Mirko Bonadei
· 7 years ago
90bace0
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
by henrika
· 7 years ago
d79314f
Reland "Add fine grained dropped video frames counters on sending side"
by Ilya Nikolaevskiy
· 7 years ago
1c1a681
Revert "Add fine grained dropped video frames counters on sending side"
by Ilya Nikolaevskiy
· 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
4b1a363
Add fine grained dropped video frames counters on sending side
by Ilya Nikolaevskiy
· 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
05d9822
Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss.
by Taylor Brandstetter
· 7 years ago
b709cf8
Remove Call::ParseRtpPacket
by Danil Chapovalov
· 7 years ago
245660a
Fix Gn untracked headers in webrtc/call.
by Mirko Bonadei
· 7 years ago
4bece9a
Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled.
by Åsa Persson
· 7 years ago
4332d09
Reland "Reland "Remove WEBRTC_TRACE.""
by Fredrik Solenberg
· 7 years ago
a32dd01
Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
by Fredrik Solenberg
· 7 years ago
39cefdb
Revert "Reland "Remove WEBRTC_TRACE.""
by Fredrik Solenberg
· 7 years ago
68007e9
Reland "Remove WEBRTC_TRACE."
by Fredrik Solenberg
· 7 years ago
4a87e1c
Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead
by Elad Alon
· 7 years ago
729b910
Revert "Remove WEBRTC_TRACE."
by Fredrik Solenberg
· 7 years ago
2209b90
Remove WEBRTC_TRACE.
by Fredrik Solenberg
· 7 years ago
d4404c2
Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
by Fredrik Solenberg
· 7 years ago
34cdd2d
Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
by Fredrik Solenberg
· 7 years ago
b0a0207
Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
by Gustaf Ullberg
· 7 years ago
440216f
Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets.
by Bjorn Terelius
· 7 years ago
3102734
Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)."
by Rasmus Brandt
· 7 years ago
2666cf7
Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld).
by Rasmus Brandt
· 7 years ago
5aea38c
Disabling CallPerfTest.{CaptureNtpTimeWithNetworkDelay,CaptureNtpTimeWithNetworkJitter}.
by Alex Loiko
· 7 years ago
06319b7
Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Mac.
by Alex Loiko
· 7 years ago
1405afe
Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Linux due to flakiness.
by lliuu
· 7 years ago
3b3622f
Delete member VideoReceiveStream::Config::Rtp::ulpfec.
by nisse
· 7 years ago
2c30120
Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ )
by brandtr
· 7 years ago
2cefac6
Add full stack tests for MediaCodec encoder.
by brandtr
· 7 years ago
99a81b6
Remove #include of rtc_stream_config.h from rtc_event_log.h
by Elad Alon
· 7 years ago
9a2e906
Added RTCMediaStreamTrackStats.concealmentEvents
by Gustaf Ullberg
· 7 years ago
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 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