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gerrit-public.fairphone.software
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platform
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external
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webrtc
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3c938fc5ea57d4a90dbc550eb235bbff53a079f7
3c938fc
Add NetEq delay plotting to event_log_visualizer
by henrik.lundin
· 7 years ago
3c81a1a
Add field trial for balanced degradation preference.
by asapersson
· 7 years ago
c417d9e
NetEq: Removing LastError and LastDecoderError
by Henrik Lundin
· 7 years ago
2b3aa14
Fix Chromium style checker warnings for MockAudioDecoder
by kwiberg
· 7 years ago
96444ae
Implement operator<< for AudioCodecInfo and AudioCodecSpec
by kwiberg
· 7 years ago
6c4ba9f
Plot acknowledged bitrate when compiled with rtc_enable_bwe_test_logging.
by terelius
· 7 years ago
58c5a7d
Revert of Roll chromium_revision 05ba7bc226..78764cfda4 (479231:479277) (patchset #1 id:1 of https://codereview.webrtc.org/2936153002/ )
by mbonadei
· 7 years ago
f7e294d
Implement kBalanced degradation preference.
by asapersson
· 7 years ago
b749e5e
Fix for broken test BweFeedbackTest.
by tschumim
· 7 years ago
b7d6015
Roll chromium_revision 05ba7bc226..78764cfda4 (479231:479277)
by buildbot
· 7 years ago
7dce727
Roll chromium_revision 53ecf9341f..05ba7bc226 (479165:479231)
by buildbot
· 7 years ago
6eb03b8
Remove dependency on gunit headers in virtualsocketserver.
by Bjorn Mellem
· 7 years ago
1ee2125
Adding PortAllocator option to support cases where sockets can't be bound.
by deadbeef
· 7 years ago
1d560e1
Roll chromium_revision 4ddaa6f836..53ecf9341f (479034:479165)
by buildbot
· 7 years ago
179f997
Remove DCHECK from PeerConnectionFactory::worker_thread.
by zstein
· 7 years ago
da4eba1
Tune vp9 quality scaler parameters
by glaznev
· 7 years ago
0c61a36
Roll chromium_revision 4f7c2dc196..4ddaa6f836 (478995:479034)
by buildbot
· 7 years ago
5c4eebb
Implement org.webrtc.VideoEncoder using the android MediaCodec.
by Bjorn Mellem
· 7 years ago
7be7883
Adds detection of audio glitches for playout on iOS (reland)
by henrika
· 7 years ago
6e286cb
Revert "Adds detection of audio glitches for playout on iOS. "
by Henrik Andreasson
· 7 years ago
33e4e65
Adds detection of audio glitches for playout on iOS.
by henrika
· 7 years ago
dea075c
Log an error in RtpDemuxer::FindSsrcAssociations() if kMaxProcessedSsrcs exceeded
by eladalon
· 7 years ago
7ed35f4
Replacing WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP with WEBRTC_ENABLE_PROTOBUF.
by minyue-webrtc
· 7 years ago
10e1f75
Roll chromium_revision 9061a92f5c..4f7c2dc196 (478958:478995)
by buildbot
· 7 years ago
2986033
Remove webrtcvideoengine2.h
by eladalon
· 7 years ago
659a010
Delete old include file webrtc/video_frame.h.
by nisse
· 7 years ago
a65ad22
Delete unused method FilesystemInterface::GetFileTime.
by nisse
· 7 years ago
8c6afef
Make sure UI methods get called on the main thread
by adam.fedor
· 7 years ago
fdfeb83
Declaring rtc_base_approved dep on webrtc_common
by mbonadei
· 7 years ago
7339712
Removing backward compatible header
by mbonadei
· 7 years ago
a735d4e
Roll chromium_revision 0ca6ede735..9061a92f5c (478917:478958)
by buildbot
· 7 years ago
2c9f9f2
Only create H264 frames if there are no gaps in the packet sequence number.
by philipel
· 7 years ago
fc30975
Access UIApplication on main thread
by Anders Carlsson
· 7 years ago
5b383c0
Revert "Update webrtc/sdk/objc to new VideoFrameBuffer interface"
by Magnus Jedvert
· 7 years ago
1edbda0
Don't hardcode gn target path for licence generation.
by Kári Tristan Helgason
· 7 years ago
f3ba648
Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format
by Danil Chapovalov
· 7 years ago
29f0d45
Delete ApplicationName and OrganizationName.
by nisse
· 7 years ago
b008b45
Update webrtc/sdk/objc to new VideoFrameBuffer interface
by Magnus Jedvert
· 7 years ago
687bc3e
Delete unused method Win32Filesystem::GetAppPathname.
by nisse
· 7 years ago
418b7d3
Increase number of unsignaled audio streams we handle to 4.
by solenberg
· 7 years ago
c18c49b
Roll chromium_revision 239d4798df..0ca6ede735 (478894:478917)
by buildbot
· 7 years ago
f52ef71
Delete unused method FilesystemInterface::DeleteEmptyFolder.
by nisse
· 7 years ago
f9fc4a5
Roll chromium_revision 97580dea94..239d4798df (478848:478894)
by buildbot
· 7 years ago
385a6e4
Roll chromium_revision 15b2b0b0e9..97580dea94 (478791:478848)
by buildbot
· 7 years ago
c35c7de
Fix play block size mismatch in Win audio device.
by lliuu
· 7 years ago
84da736
Roll chromium_revision 71baf2eb8f..15b2b0b0e9 (478645:478791)
by buildbot
· 7 years ago
22e0814
Update VirtualSocketServerTest to use a fake clock.
by deadbeef
· 7 years ago
36b1a5f
Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:
by yujo
· 7 years ago
0703856
Add SafeClamp(), which accepts args of different types
by kwiberg
· 7 years ago
d1114c7
Roll chromium_revision d59edeefb6..71baf2eb8f (478597:478645)
by buildbot
· 7 years ago
38018ba
Merge BitrateControllerImpl::RtcpBandwidthObserverImpl into BitrateControllerImpl
by Danil Chapovalov
· 7 years ago
42742a5
Fall-back to OpenGL renderer if mac hardware doesn't support Metal
by adam.fedor
· 7 years ago
84b4d2c
Use rtp_header_extension_map.h instead of rtp_header_extension.h
by Danil Chapovalov
· 7 years ago
d3d8702
Roll chromium_revision 6dcccd8c3f..d59edeefb6 (478515:478597)
by buildbot
· 7 years ago
7f8369a
Update expectation of OneBitrateObserverTwoRtcpObservers test:
by Danil Chapovalov
· 7 years ago
f474c19
ACM tests: separate checksums for Android ARM64 clang and non-clang
by Henrik Lundin
· 7 years ago
39a41d9
Split rtc_task_queue target. Add separate target for sequenced_task_checker and weak_ptr.
by perkj
· 7 years ago
7123029
List all device resolutions in AppRTCMobile settings
by Anders Carlsson
· 7 years ago
c276ecf
Update Android video buffers to new VideoFrameBuffer interface
by Magnus Jedvert
· 7 years ago
f184138
s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine
by eladalon
· 7 years ago
a8e781a
Make rtc_event_log2text output header extensions
by ilnik
· 7 years ago
3fae628
Reland Refactored incoming bitrate estimator.
by tschumim
· 7 years ago
90e3190
Update webrtc/test to new VideoFrameBuffer interface
by Magnus Jedvert
· 7 years ago
72dbe2a
Revert "Revert "Update video_coding/codecs to new VideoFrameBuffer interface""
by Magnus Jedvert
· 7 years ago
29584c5
Roll chromium_revision 4b325fbec4..6dcccd8c3f (478514:478515)
by buildbot
· 7 years ago
ef0a3ea
Roll chromium_revision 5a101abbe0..4b325fbec4 (478513:478514)
by buildbot
· 7 years ago
c8cac10
Roll chromium_revision 632b145c0e..5a101abbe0 (478512:478513)
by buildbot
· 7 years ago
7e120eb
Roll chromium_revision 8e89b0b1a1..632b145c0e (478506:478512)
by buildbot
· 7 years ago
d0fa397
Roll chromium_revision 999a40e458..8e89b0b1a1 (478482:478506)
by buildbot
· 7 years ago
ad3a029
Roll chromium_revision 1b59498f08..999a40e458 (478431:478482)
by buildbot
· 7 years ago
995bad0
Roll chromium_revision 524fdc6e30..1b59498f08 (478357:478431)
by buildbot
· 7 years ago
c131bf9
Enable webrtc_nonparallel_tests on iOS simulator
by kjellander
· 7 years ago
b82487b
Roll chromium_revision f7c1799c98..524fdc6e30 (478294:478357)
by buildbot
· 7 years ago
be767e0
Remove default impl of Attach/DetachAecDump.
by Alex Loiko
· 7 years ago
12149bd
Roll chromium_revision 06a62c1231..f7c1799c98 (478256:478294)
by buildbot
· 7 years ago
76d29f9
Fix Channel::GetSendCodec when used together with SetEncoder.
by ossu
· 7 years ago
7fdd067
Roll chromium_revision f8c224c31c..06a62c1231 (478239:478256)
by buildbot
· 7 years ago
461c940
ObjC: Rename VideoToolbox/decoder.cc to VideoToolbox/decoder.mm
by Magnus Jedvert
· 7 years ago
b4ab381
Use the configured remote ssrc instead of relying on the first received packet RtpStreamReceiver.
by stefan
· 7 years ago
fee994c
Ensure the openGLContext is current before trying to reshape the viewport
by adam.fedor
· 7 years ago
b1f2ff9
Rename class RtpStreamReceiver --> RtpVideoStreamReceiver.
by nisse
· 7 years ago
e2baffb
Create a UIApplication when running tests on iOS.
by Kári Tristan Helgason
· 7 years ago
fae6d09
Roll chromium_revision 74ece38823..f8c224c31c (478141:478239)
by buildbot
· 7 years ago
85dcaea
Roll chromium_revision 423c0eff45..74ece38823 (478099:478141)
by buildbot
· 7 years ago
6baee78
Add missing #include <cerrno> in string_to_number.cc
by hugoh
· 7 years ago
46537a3
Avoiding cascaded software echo cancellers
by Per Åhgren
· 7 years ago
7412fe6
Roll chromium_revision 2108fde0a1..423c0eff45 (478041:478099)
by buildbot
· 7 years ago
dc4f7f5
Roll chromium_revision 88476a9f88..2108fde0a1 (477979:478041)
by buildbot
· 7 years ago
59154ed
Roll chromium_revision 61a28216c8..88476a9f88 (477949:477979)
by buildbot
· 7 years ago
20e4a73
MockAecDump and integration tests between AecDump and AudioProcessing
by aleloi
· 7 years ago
317005a
Revert of Periodically update codec bit/frame rate settings. (patchset #2 id:160001 of https://codereview.webrtc.org/2924023002/ )
by sprang
· 7 years ago
d3a8119
Roll chromium_revision 53a49d4c81..61a28216c8 (477934:477949)
by buildbot
· 7 years ago
cf705c5
Reland of Protect new header extension by field trial experiment to allow hardcoding it in SDP (patchset #1 id:1 of https://codereview.webrtc.org/2922723002/ )
by ilnik
· 7 years ago
cdafeda
Reland of Periodically update codec bit/frame rate settings. (patchset #1 id:1 of https://codereview.webrtc.org/2923993002/ )
by sprang
· 7 years ago
1066b13
Remove deprecated AudioMixerImpl creation method.
by Alex Loiko
· 7 years ago
d0244c2
Add RSID-based demuxing to RtpDemuxer
by eladalon
· 7 years ago
5c4897f
Roll chromium_revision 3c550cc859..53a49d4c81 (477916:477934)
by buildbot
· 7 years ago
15dcb38
Make error resilience configurable through VideoCodecVP9 resilience setting (removes hard coded value in vp9_impl.cc).
by asapersson
· 7 years ago
04ca637
Make 'aleloi@' OWNER of webrtc/modules/audio_processing
by Alex Loiko
· 7 years ago
75b68b9
Delete webrtc/call.h (replaced with webrtc/call/call.h).
by nisse
· 7 years ago
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