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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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ae70876c00ac644bcef8c274e1f12162f9b936d3
ae70876
Remove unnecessary styling for some controls in ARDMainView.m for ios.
by denicija
· 8 years ago
d17d536
Add setting to AppRTCMobile for iOS, that can change capture resolution.
by denicija
· 8 years ago
b763e39
Add CreateWindowCapturer() and CreateScreenCapturer() in DesktopCapturer
by zijiehe
· 8 years ago
ee8ad2b
Adding data channel ID to Java binding of DataChannel.
by deadbeef
· 8 years ago
8a44e1d
Let RTC_[D]CHECK_op accept arguments of different signedness
by kwiberg
· 8 years ago
803d97f
Let ViEEncoder express resolution requests as Sinkwants.
by perkj
· 8 years ago
b1ed609
Use rtcp::Bye instead of RTCPUtility parser for rtcp_sender_unittest
by danilchap
· 8 years ago
a27172d
Adding audio only mode to video loopback test.
by minyue
· 8 years ago
673383b
CQ: Add Android and Linux "more configs" bots
by Henrik Kjellander
· 8 years ago
384e731
vp8_impl.cc: Adjust cpu speed setting for arm for devices with 4 or more cores.
by asapersson
· 8 years ago
91d96aa
Add third_party/android_support_test_runner to .gitignore
by ehmaldonado
· 8 years ago
aee3e0e
Only advance |first_seq_num_| if packets are explicitly cleared from the PacketBuffer.
by philipel
· 8 years ago
b521aa7
Clean up abs-send-time for audio.
by stefan
· 8 years ago
aca3a24
Moving stun_prober target from webrtc/p2p to webrtc/examples
by charujain
· 8 years ago
eeafe94
RTCInboundRTPStreamStats[1] added.
by hbos
· 8 years ago
b84ad63
Add RTCP packet class for signaling encoder target bitrate.
by sprang
· 8 years ago
6ded190
RTCOutboundRTPStreamStats[1] added.
by hbos
· 8 years ago
15ca8f6
Let receiving() and SignalRecevingState be part of rtc::PacketTransportInterface.
by johan
· 8 years ago
fe647f4
Add ability to handle data from multiple streams in RateAccCounter.
by asapersson
· 8 years ago
7eaa836
Revert of RTCOutboundRTPStreamStats added. (patchset #3 id:80001 of https://codereview.webrtc.org/2456463002/ )
by perkj
· 8 years ago
4ed0750
Revert of RTCInboundRTPStreamStats added. (patchset #4 id:100001 of https://codereview.webrtc.org/2452043002/ )
by perkj
· 8 years ago
0d7bf16
RTCInboundRTPStreamStats[1] added.
by hbos
· 8 years ago
69e9cb0
RTCOutboundRTPStreamStats[1] added.
by hbos
· 8 years ago
bb9212a
Add ffmpeg and zxing to webrtc/tools/video_quality_toolchain.
by Henrik Kjellander
· 8 years ago
e566ac7
Remove voe::Channel::StopReceive() and associated logic.
by solenberg
· 8 years ago
9c41e47
Remove unnecessary test fixture in codec_unittest.cc
by magjed
· 8 years ago
6b6c88f
NetEq jitter calculation now done in uint64_t.
by ossu
· 8 years ago
80ac24d
Allow max 1 block per type in RTCP Extended Reports
by danilchap
· 8 years ago
ba156cf
Improvements in how WebRTC.Audio.RecordedOnlyZeros is added as histogram.
by henrika
· 8 years ago
67dca9f
Delete ShallowCopy, in favor of copy construction and assignment.
by nisse
· 8 years ago
c846f2f
Fix out_frame argument of PreprocessFrameAndVerify.
by nisse
· 8 years ago
87da404
Implement qpSum stat for video send ssrc stats.
by sakal
· 8 years ago
fffc1e5
Add functionality for parsing H264 profile-level-id
by magjed
· 8 years ago
f0a7c5a
Delete deprecated method VideoFrame::CreateFrame.
by nisse
· 8 years ago
626bc95
Reland of "Separating video settings in VideoQualityTest".
by minyue
· 8 years ago
869e7cd
Rename ProducerFec to UlpfecGenerator.
by brandtr
· 8 years ago
d55c3f6
Rename FecReceiver to UlpfecReceiver.
by brandtr
· 8 years ago
6b825df
Using AudioOption to enable audio network adaptor.
by minyue
· 8 years ago
4ee7046
Add unit tests for bandwidth limited resolution stats in SendStatisticsProxy.
by asapersson
· 8 years ago
535830e
Rename Fec to Ulpfec in EndToEndTests.
by brandtr
· 8 years ago
ca27f9d
It seems that if encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceNum is configured to number of cores as determined by openh264 (or any number > 1 in my local tests), frame rate statistics will be mucked up (apparently thousands of frames per second) and quality will bottom out because bits per frame is then very low.
by sprang
· 8 years ago
e602f0a
Rename Fec to Ulpfec in VideoSendStreamTest.
by brandtr
· 8 years ago
42ca68a
Ensure one does not register same rtp header extension with different id
by danilchap
· 8 years ago
051f678
Add a NeededFrequency() method to the AudioMixer::Source interface.
by aleloi
· 8 years ago
9aa7883
Revert of "Separating video settings in VideoQualityTest". (patchset #4 id:60001 of https://codereview.webrtc.org/2314403007/ )
by minyue
· 8 years ago
bc80744
Eliminate left shift of negative value by using multiplication instead
by kwiberg
· 8 years ago
c5b435d
Re-enable PostDelayed test for TaskQueue on Windows.
by tommi
· 8 years ago
a144be3
Delete videorendererfactory.h and cricket::GdiVideoRenderer.
by nisse
· 8 years ago
18b8774
Setting PATH so that the 'plistbuddy' utility can be found, in a typical OS X environment
by VladimirTechMan
· 8 years ago
21d45d2
Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known.
by Per
· 8 years ago
b68d655
Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions
by zijiehe
· 8 years ago
fcab7d6
Revert of Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions (patchset #3 id:120001 of https://codereview.chromium.org/2452263003/ )
by zijiehe
· 8 years ago
9cb0b3b
Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions
by zijiehe
· 8 years ago
05a55b5
Revert of Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #2 id:20001 of https://codereview.webrtc.org/2455963004/ )
by emircan
· 8 years ago
e7fc7d5
Fixing flaky DtmfSenderTest by using fake clock.
by deadbeef
· 8 years ago
3e9a537
Original CL: https://codereview.webrtc.org/2433153003/, commit 8b8d3e4c30e8ea3846b58dfd36d1fd35a7799df4.
by ivoc
· 8 years ago
1e45cc6
Replace WebRtcVideoEncoderFactory::VideoCodec with cricket::VideoCodec
by magjed
· 8 years ago
e2a0177
Style cleanups in rtp header extension traits:
by danilchap
· 8 years ago
af27ed0
Add algorithm for Residual Echo Detector.
by ivoc
· 8 years ago
5f1b051
Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known.
by perkj
· 8 years ago
c4b9b94
Revert of Replace FileWrapper with File (in audio_device) (patchset #3 id:40001 of https://codereview.webrtc.org/2386963003/ )
by terelius
· 8 years ago
6ceab08
GN: New conventions, default target and refactorings
by kjellander
· 8 years ago
9f4a4a0
Add empty residual echo detector.
by ivoc
· 8 years ago
a9a1ac2
Moved rtc::Base64 to base approved.
by philipel
· 8 years ago
02ba211
Move RTCStatsCollector helper functions to anonymous namespace.
by hbos
· 8 years ago
f005a00
Added calling of the stream_analog_level api in audioproc_f
by peah
· 8 years ago
6d6762c
Add UINavigationController and settings bar button to AppRTCMobile.
by denicija
· 8 years ago
5da65f2
Revert of Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #4 id:60001 of https://codereview.webrtc.org/2455063002/ )
by perkj
· 8 years ago
48dfab5
Revert of New statistics interface for APM (patchset #11 id:200001 of https://codereview.webrtc.org/2433153003/ )
by ivoc
· 8 years ago
461c29e
Change ViEEncoder to not reconfigure the encoder until the video resolution is known.
by perkj
· 8 years ago
135259a
In order to be able to analyze the AGC behavior on
by peah
· 8 years ago
04055e9
Removes all uses of FileWrapper in audio_device.
by palmkvist
· 8 years ago
8b8d3e4
New statistics interface for APM
by ivoc
· 8 years ago
37abf53
Delete FrameObject::size member.
by nisse
· 8 years ago
11f72b1
Fix compile error for non Intel platforms
by Gordana.Cmiljanovic
· 8 years ago
9922016
Fix "IsLoopbackIp" to cover all loopback addresses; not just 127.0.0.1.
by deadbeef
· 8 years ago
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
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