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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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6be0a657c563acfb3f9b2502c4453d7ba46ef0d1
6be0a65
Move ScreenCapturer 'real' tests out of screen_capturer_unittest.cc.
by zijiehe
· 8 years ago
32bcaf6
Improve RTC_DCHECK_op so that it won't trigger useless compiler warnings
by kwiberg
· 8 years ago
a73df55
Do not rely on specific ordering on generated candidates in TestGetAllPortsPortRange
by honghaiz
· 8 years ago
45d18eb
Re-enable the PostDelayed TaskQueue test on all platforms except Windows.
by tommi
· 8 years ago
492ee28
Use bayesian estimate of acked bitrate.
by Stefan Holmer
· 8 years ago
9890a58
Testing of FileVideoCapturer.
by mandermo
· 8 years ago
da35f3e
Delete unused features of rtc::FilesystemInterface and related classes.
by nisse
· 8 years ago
a101e56
Remove LOGGING=1 define.
by kjellander
· 8 years ago
fe90b41
Improves audio logs of native audio layers on Android
by henrika
· 8 years ago
68e6bdd
Remove use of VoECodec in video/call tests.
by solenberg
· 8 years ago
5e49c2f
Restore symbol level for Android builds.
by kjellander
· 8 years ago
bc59b06
Add gn_isolate_map.pyl file for WebRTC stand-alone tests
by kjellander
· 8 years ago
b112568
Roll chromium_revision 9b5bb47fa0..04e7c673d9 (426837:427632)
by kjellander
· 8 years ago
e183121
Enable clang style plugin in webrtc/modules/desktop_capture
by sergeyu
· 8 years ago
54b0acb
Change destruction order to fix potential invalid pointer dereference.
by erikchen
· 8 years ago
489c0d4
Decrease threshold for key frame generation.
by glaznev
· 8 years ago
91c2d43
Disable TaskQueueTest.PostDelayed because of flakiness
by terelius
· 8 years ago
e5ba44e
Implement framesDecoded stat in video receive ssrc stats.
by sakal
· 8 years ago
784a831
Check that stats_proxy_ is non-NULL before use.
by nisse
· 8 years ago
5819660
MB: Add Linux swarming bots with memory sanitizers on the FYI waterfall.
by ehmaldonado
· 8 years ago
059fb44
- Replace FakeAudioProcessing in WVoE unittest with MockAudioProcessing.
by solenberg
· 8 years ago
16b6d6d
Reland of "Separating video settings in VideoQualityTest".
by minyue
· 8 years ago
c1600c5
Follow standard sending CVO rtp header extension
by danilchap
· 8 years ago
b906172
Reland of Move bitstream parser to more appropriate directory. (patchset #1 id:1 of https://codereview.webrtc.org/2430353004/ )
by kthelgason
· 8 years ago
12ba186
Move parsing from tests to Transport helper in RTPSenderTests
by danilchap
· 8 years ago
a8bec8d
Testing of VideoFileRenderer with byte frames
by mandermo
· 8 years ago
940b6d6
Clean up logging in AudioSendStream::SetupSendCodec().
by solenberg
· 8 years ago
da389e3
PrintTo functions for RTCStats added in rtcstatscollector_unittest.cc
by hbos
· 8 years ago
d89ab14
Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit.
by johan
· 8 years ago
57e13de
Minor cleanup of rtc::BasicPacketSocketFactory implementation.
by johan
· 8 years ago
0d8ade5
Remove remnants of libsrtp1
by mattdr
· 8 years ago
257dc39
Refactoring: Hide VideoCodec.codecSpecific as "private"
by hta
· 8 years ago
189f9b1
Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ )
by terelius
· 8 years ago
d0af5c6
Fix a deadlock in EglRenderer.releaseEglSurface.
by sakal
· 8 years ago
2d81eb3
Fix BWE simulations so that it uses the delay based BWE.
by terelius
· 8 years ago
1836fd6
Clean up logging in AudioSendStream::SetupSendCodec().
by solenberg
· 8 years ago
701d628
Moved the AGC render sample queue into the audio processing module
by peah
· 8 years ago
d8872c5
Removed the file resources/audioproc.aecdump.sha1 file which is no longer used.
by peah
· 8 years ago
a062460
Several subcomponents inside APM copy render audio from
by peah
· 8 years ago
67c8bc4
RTCStats equality operator added.
by hbos
· 8 years ago
01bbc3c
Reland of Deflake ChangingNetworkRoute test.
by stefan
· 8 years ago
77f5953
Revert of Deflake ChangingNetworkRoute test. (patchset #1 id:1 of https://codereview.webrtc.org/2426073002/ )
by ehmaldonado
· 8 years ago
6711820
Deflake ChangingNetworkRoute test.
by stefan
· 8 years ago
cc34833
Remove now unused code in RtpHeaderExtensionMap
by danilchap
· 8 years ago
611f267
Make WebRTC compatible with OpenH264 v1.6.
by sprang
· 8 years ago
af1ae31
Remove dead dependencies on xmllite and xmpp.
by kjellander
· 8 years ago
dbf6705
codec_unittest.cc: Fix TEST vs TEST_F mismatch
by Magnus Jedvert
· 8 years ago
06c8e1e
Revert of H264 codec: Check profile-level-id when matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ )
by Magnus Jedvert
· 8 years ago
fcba8fe
Delete left-over file profiler_unittest.cc.
by nisse
· 8 years ago
74097fd
Delete unused file screencastid.h.
by nisse
· 8 years ago
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
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