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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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e58d73d23ea7d3338c3b4617943224f9c71a2612
e58d73d
Fix more swarming test failures by using the fake clock or longer timeout.
by honghaiz
· 8 years ago
a6b8298
Use relative names in GN to make Chromium happy
by kwiberg
· 8 years ago
4a18f16
Update XServerPixelBuffer to handle errors returned from XGetImage().
by sergeyu
· 8 years ago
da2bf4e
Stop using old AudioCodingModule::RegisterReceiveCodec overloads
by kwiberg
· 8 years ago
88b7074
Remove unused function implementations from FakeWebRtcVoiceEngine.
by solenberg
· 8 years ago
fb70b45
Preventing TURN redirects to loopback addresses.
by deadbeef
· 8 years ago
838cdb3
Revert of Fix chromium-style warnings. (patchset #1 id:1 of https://codereview.webrtc.org/2400993002/ )
by terelius
· 8 years ago
5d79a7c
rtcstats_objects.h updated with TODOs about stats not being collected
by hbos
· 8 years ago
a6f495c
Simplifying audio network adaptor by moving receiver frame length range to ctor.
by minyue
· 8 years ago
a73f6c9
NetEq now works with packets as values, rather than pointers.
by ossu
· 8 years ago
d312713
Roll chromium_revision 1362287708..9b5bb47fa0 (426760:426837) + roll Android SDK to N
by ehmaldonado
· 8 years ago
86b92e0
Drop VP8 frames older than the last sync frame in the RtpFrameReferenceFinder.
by philipel
· 8 years ago
1655e45
Elimiteted race condition in the AudioMixer.
by aleloi
· 8 years ago
2206c95
Revert of Fix some chromium style warnings in remote_bitrate_estimator.h (patchset #1 id:1 of https://codereview.webrtc.org/2387113008/ )
by terelius
· 8 years ago
0140408
Add tests and fix thread annotations
by danilchap
· 8 years ago
b60d196
Eliminate left shift of negative value by using multiplication instead
by kwiberg
· 8 years ago
2fa7c67
RTCTransportStats[1] added, supporting all members.
by hbos
· 8 years ago
5de3a7e
Remove unused variable from delay based BWE.
by terelius
· 8 years ago
509eadd
Fix chromium-style warnings.
by terelius
· 8 years ago
c22bcf4
Fix some chromium style warnings in remote_bitrate_estimator.h
by terelius
· 8 years ago
d7ce668
Roll chromium_revision f9e01d4887..1362287708 (426685:426760)
by buildbot
· 8 years ago
a3cac05
GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn
by kjellander
· 8 years ago
43536c3
Implement framesEncoded stat in video send ssrc stats.
by sakal
· 8 years ago
2675274
Remove cricket::VideoCodec with, height and framerate properties
by perkj
· 8 years ago
d3c4008
Delete always-zero ByteBufferWriter::start_.
by nisse
· 8 years ago
61c053e
Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
by nisse
· 8 years ago
ebf5240
Allow using Java classes that don't require JNI in Chromium.
by sakal
· 8 years ago
66712b0
Revert of Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource. (patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ )
by nisse
· 8 years ago
151572b
Delete unused class AudioSourceWithMixStatus.
by nisse
· 8 years ago
25445d3
Integrate FlexfecReceiveStream with Call.
by brandtr
· 8 years ago
764e364
Several subcomponents inside APM copy render audio from
by peah
· 8 years ago
8974973
Added the missing ReadQueuedRenderData() call to the AECM bitexactness test
by peah
· 8 years ago
12986c4
Added the missing ReadQueuedRenderData() call to the gain controller bitexactness test
by peah
· 8 years ago
da38293
Added the missing ReadQueuedRenderData() call to the AEC bitexactness test
by peah
· 8 years ago
894c400
Android VideoFileRenderer: Wait for posted frames in release()
by Magnus Jedvert
· 8 years ago
8c63a82
Add a placeholder stat for logging the estimated residual echo likelihood.
by ivoc
· 8 years ago
3355f6d
Avoids invalid copy of audio buffer to task queue.
by henrika
· 8 years ago
c4d2dc4
Delete DataLog abstraction, which was almost unused.
by nisse
· 8 years ago
dda1e60
Reland of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2441493003/ )
by nisse
· 8 years ago
bd16341
Roll chromium_revision 4c4977aa05..f9e01d4887 (426117:426685)
by buildbot
· 8 years ago
84fbf9e
SUCCEEDED macro is misused
by zijiehe
· 8 years ago
bdb8df8
BringSelectedWindowToFront should bring the window to front instead of only focusing it
by zijiehe
· 8 years ago
97abf24
Use variadic templates instead of pump for RefCountedObject
by danilchap
· 8 years ago
6c27849
Move audio frame memory handling inside AudioMixer.
by aleloi
· 8 years ago
920d30b
Replaced thread checker with race checker in AudioMixer.
by aleloi
· 8 years ago
161a586
Fix some flaky tests by using longer timeout and/or fake clock.
by Honghai Zhang
· 8 years ago
b9eaeba
Return nullptr from RTCCertificate::FromPEM on failure.
by jbroman
· 8 years ago
58000a0
Move shared_desktop_frame.cc to webrtc/modules/desktop_capture:primitives
by Sergey Ulanov
· 8 years ago
142f019
Append second nack list in same compound rtcp packet instead of replace
by danilchap
· 8 years ago
aed581a
Made AudioReceiveStream a mixer participant.
by aleloi
· 8 years ago
5f70d3b
Fix org.mockito.Matchers deprecation warnings in DirectRTCClientTest.
by sakal
· 8 years ago
201dfe9
Split audio mixer into interface and implementation.
by aleloi
· 8 years ago
76648da
Add FlexfecReceiveStream.
by brandtr
· 8 years ago
057b8d9
Remove all traces of Dr Memory.
by Henrik Kjellander
· 8 years ago
69034df
Make GN build screenshare_loopback
by palmkvist
· 8 years ago
5a87245
iOS: Optimize video scaling and cropping
by magjed
· 8 years ago
7a97344
Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.
by minyue
· 8 years ago
1cb4823
Android YuvConverter: Use OpenGL Framebuffer instead of EGL pixel buffer
by magjed
· 8 years ago
9ab8a18
Android: Extend functionality of EglRenderer
by magjed
· 8 years ago
ca20e7c
Revert of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2437703002/ )
by nisse
· 8 years ago
c1f8ecb
Remove check for numberOfCameras from AppRTC Mobile PeerConnectionClient.
by sakal
· 8 years ago
be4aff7
Suppress deprecation warning in CallFragment.
by sakal
· 8 years ago
aff9ff0
Create .git-blame-ignore-revs and add Java format CL to it.
by sakal
· 8 years ago
e33c5d9
Added a level controller initialization value to MediaConstraints.
by aleloi
· 8 years ago
647915f
Add loopback option and improve UX of AppRTCMobile for Mac.
by denicija
· 8 years ago
725e212
Prevent stripping of C interfaces in framework
by kthelgason
· 8 years ago
e037060
Add to rtc::Optional equality/unequality comparisions with object
by danilchap
· 8 years ago
a34e796
Delete unused file mediacommon.h.
by nisse
· 8 years ago
55928fe
QualityScaler reset bugfix
by kthelgason
· 8 years ago
0489e49
Change RefCountedObject to use perfect forwarding.
by perkj
· 8 years ago
79f0bf3
A variable in ScreenCapturerWinDirectx has a bad name
by zijiehe
· 8 years ago
f04f14e
Revert of Move bitstream parser to more appropriate directory. (patchset #4 id:60001 of https://codereview.webrtc.org/2370853005/ )
by kthelgason
· 8 years ago
cc6817e
Move current bitstream parser to more appropriate directory.
by kthelgason
· 8 years ago
577bc19
Android: Move YuvConverter to its own file
by Magnus Jedvert
· 8 years ago
b6f1fb5
Delete RTPSender::BuildRtpHeader function and all dependencies
by danilchap
· 8 years ago
061ea0d
Remove VideoCodec resolution validation.
by Per
· 8 years ago
e3e411a
Removed perkj@ from video WATCHLIST
by Per
· 8 years ago
73c5d4a
Include ScreenCapturerAndroid in libjingle_peerconnection_java.jar
by sakal
· 8 years ago
0934785
Reland of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2402853002/ )
by nisse
· 8 years ago
4e52386
Reland of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #1 id:1 of https://codereview.webrtc.org/2427733002/ )
by brandtr
· 8 years ago
249beee
Remove DesktopRegion parameter from DesktopCapturer::Capture
by zijiehe
· 8 years ago
e8295fc
Roll chromium_revision eb9b71b64b..4c4977aa05 (426008:426117)
by buildbot
· 8 years ago
6a4607e
Deflaky ScreenCapturerTest
by zijiehe
· 8 years ago
1eb1293
Handle BW drop in ALR region and initiate probing
by Irfan Sheriff
· 8 years ago
a9c7cfa
Prepare for introduction of rtc::PacketTransportInterface.
by johan
· 8 years ago
1203066
Compilerwarning possible loss of data in file port.h
by bertholdherrmann08
· 8 years ago
cc555c5
RTCDataChannelStats[1] added, supporting all stats members.
by hbos
· 8 years ago
1394c7b
Fix for flaky test: EndToEndTest.VerifyHistogramStatsWithRtx
by asapersson
· 8 years ago
0b7be9c
Roll chromium_revision c8b7ee41e0..eb9b71b64b (425645:426008)
by buildbot
· 8 years ago
9960bb1
Call OnTransportFeedback just when feedback_observer exist.
by michaelt
· 8 years ago
53fe19d
Set min and max rate on caller and on callee side.
by michaelt
· 8 years ago
64e1a32
Second try to get "Support for video file instead of camera and output video out to file" accepted
by mandermo
· 8 years ago
67a8c98
Revert of Support for video file instead of camera and output video out to file (patchset #17 id:320001 of https://codereview.webrtc.org/2273573003/ )
by kjellander
· 8 years ago
f33970b
Add unittest for I420Buffer::Rotate.
by nisse
· 8 years ago
6ed592d
Rename variables to reflect that DelayBasedBwe lives on the send side rather than receive side.
by terelius
· 8 years ago
5588a13
Now uses rtc::Buffer in AudioDeviceBuffer.
by henrika
· 8 years ago
4466699
Support for video file instead of camera and output video out to file
by mandermo
· 8 years ago
9e83c97
Add rtc::Optional::emplace
by danilchap
· 8 years ago
7a37761
Removed RTPHeader from NetEq's Packet struct.
by ossu
· 8 years ago
553024a
During a fix of an unrelated issue, a bug was introduced in the rtp analyzer tool: when the number of data points was divisible by RTPStatitstics.PLOT_RESOLUTION_MS (which is 50), pyplot.plot was called with arrays of different lengths. One of the arrays could be one element larger.
by aleloi
· 8 years ago
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