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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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13d38fbe904cd99acebf77068e93df34725eeb47
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a6a699a
Sent bitrate stats are incorrect if FlexFEC is configured:
by asapersson
· 8 years ago
6b272c5
RtpReceiver: Add RegisterReceivePayload function for VideoCodec
by magjed
· 8 years ago
5de9b6a
Move helpers_ios.cc/.h
by solenberg
· 8 years ago
0928a3c
Reland of Split out target rtc_media_base from rtc_media (patchset #1 id:1 of https://codereview.webrtc.org/2508163002/ )
by magjed
· 8 years ago
33c81d0
Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ )
by magjed
· 8 years ago
69b627d
Move smoothing filter to common audio and exp_filter to base/analytics.
by minyue
· 8 years ago
b881254
Remove RTPPayloadStrategy and simplify RTPPayloadRegistry
by magjed
· 8 years ago
56124bd
Send audio and video codecs to RTPPayloadRegistry
by magjed
· 8 years ago
b7374db
Fix parsing padding byte in rtp header extension
by danilchap
· 8 years ago
bf67663
Rename "Audio playout level" to "Audio level" on the Y-axis of the event log graph.
by ivoc
· 8 years ago
3c3aef4
Revert of Reland "Move smoothing filter to common audio". (patchset #5 id:100001 of https://codereview.webrtc.org/2520003005/ )
by minyue
· 8 years ago
223641f
Reland "Move smoothing filter to common audio".
by minyue
· 8 years ago
b365b80
Revert of Modify the paths of the resource files to point to chromium/src/tools/... (patchset #1 id:1 of https://codereview.webrtc.org/2528893002/ )
by ehmaldonado
· 8 years ago
d8ae20b
Modify the paths of the resource files to point to chromium/src/tools/...
by ehmaldonado
· 8 years ago
3cfb3ef
Added a perf test for the residual echo detector.
by ivoc
· 8 years ago
37a2111
Increase the threshold for RunPlayoutAndRecordingInFullDuplex. Again.
by ehmaldonado
· 8 years ago
3edc7f0
AGC: Add a histogram for new level
by henrik.lundin
· 8 years ago
c42d376
DataChannelInterface: Remove default implementation of methods.
by hbos
· 8 years ago
464d50f
Set rtc_use_memcheck=true for the FYI bot.
by ehmaldonado
· 8 years ago
ed8c8ed
Add rtc_use_memcheck flag, update MB and GN to handle it, and add gni files listing the runtime deps
by ehmaldonado
· 8 years ago
d44d0ba
For VPN network, use the underlying network type as its type.
by honghaiz
· 8 years ago
4dfb8ce
Make the default value of rtcp-mux policy to required.
by zhihuang
· 8 years ago
e02407a
Add myself to WATCHLIST for api/.
by solenberg
· 8 years ago
42eee12
RTCPeerConnectionStats: Removed fixed TODO comments.
by hbos
· 8 years ago
08be780
Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ )
by sakal
· 8 years ago
817208b
Re-enables AudioDeviceTest.StartStopPlayout on Android
by henrika
· 8 years ago
8b64628
Add fps reduction API to SurfaceViewRenderer.
by sakal
· 8 years ago
4fe3b8d
Add framelistener functionality to SurfaceViewRenderer.
by sakal
· 8 years ago
1c82884
Remove binding framebuffer from GlTextureFrameBuffer.setSize.
by sakal
· 8 years ago
8e321c8
CQ: Disable android_more_configs trybot
by Henrik Kjellander
· 8 years ago
0c5a154
Try to deflake VideoSendStream tests with FlexFEC.
by brandtr
· 8 years ago
0adb828
RTCCodecStats[1] added.
by hbos
· 8 years ago
71caaca
Split avfoundationcapturer classes in separate files.
by denicija
· 8 years ago
90ea736
Add DesktopFrame rotation functions
by zijiehe
· 8 years ago
e2b1501
Start probes only after network is connected.
by Sergey Ulanov
· 8 years ago
1c062bf
Fix module/desktop_capture compilation on iOS
by Sergey Ulanov
· 8 years ago
c1dd1a5
Really disable Opus complexity tests on Android
by henrik.lundin
· 8 years ago
d661e9c
WebRTC: Replace ProjectRootPath by ResourcePath
by ehmaldonado
· 8 years ago
10165ab
Unify VideoCodecType to/from string functionality
by magjed
· 8 years ago
2d60e53
H264 encoder: Include QP information in encoded images
by magjed
· 8 years ago
e60f020
iOS AppRTCMobile: Fix SDP video codec reordering for multiple H264 profiles
by magjed
· 8 years ago
8271d04
This CL introduces the new functionality for setting
by peah
· 8 years ago
30a12fb
AGC: Add a histogram for clipping adjustment
by henrik.lundin
· 8 years ago
24d812d
DEPS: Specify WebRTC hooks and add a few dependencies
by kjellander
· 8 years ago
ab6996d
Enable QP parsing from CABAC bitstreams
by kthelgason
· 8 years ago
04c0722
Replace AudioConferenceMixer with AudioMixer.
by aleloi
· 8 years ago
b426040
Add Full HD and 4K camera resolutions to AppRTCMobile Android.
by sakal
· 8 years ago
2df1ab4
MB: Add Win32 SyzyASan (swarming) config.
by ehmaldonado
· 8 years ago
17338d4
Created an AudioMixer mock in webrtc/api/test.
by aleloi
· 8 years ago
0eb1960
ComfortNoise: Calculate used scale factor in Q13
by ossu
· 8 years ago
58f90a7
Disable Opus complexity tests on Android
by henrik.lundin
· 8 years ago
03d5fb1
Let MediaSession generate a FlexFEC SSRC when FlexFEC is active.
by brandtr
· 8 years ago
0dbb6f5
Fix the standard deviation calculation in the level controller perf tests.
by ivoc
· 8 years ago
820f578
RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video.
by hbos
· 8 years ago
468da7c
Wire up FlexFEC in VideoEngine2.
by brandtr
· 8 years ago
d848a56
DEPS: Cleanup extra_gyp_flag and extra_gitignore.py
by kjellander
· 8 years ago
875862c
Let Opus increase complexity for low bitrates
by henrik.lundin
· 8 years ago
b1e6d5e
Set surface view surface size to minimum of the layout size and frame size.
by sakal
· 8 years ago
f6acc2a
Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
by magjed
· 8 years ago
0ce6aaf
Move androidvideotracksource from api under api/android/jni.
by sakal
· 8 years ago
f723312
Add an empty libjingle_peerconnection_metrics_default_jni target.
by sakal
· 8 years ago
9688e38
Add support for FEC-FR semantics in StreamParams.
by brandtr
· 8 years ago
96385e0
iOS: Add FlexFEC-03 field trial.
by brandtr
· 8 years ago
fb94cd6
build_ios_libs.sh: Add command line bitcode option.
by tkchin
· 8 years ago
7a07f13
Fix TimeCallback used by BoringSSL.
by deadbeef
· 8 years ago
1b0e3aa
Remove deprecated CroppingWindowCapturer::Create
by zijiehe
· 8 years ago
2874796
RTCStats operator== bugfix
by hbos
· 8 years ago
f570a28
Revert of Allow custom metrics implementations on Android. (patchset #11 id:260001 of https://codereview.webrtc.org/2403463002/ )
by philipel
· 8 years ago
ab102f1
Update gtest-parallel and introduce gtest-parallel-wrapper.
by ehmaldonado
· 8 years ago
de609b2
Allow custom metrics implementations on Android.
by sakal
· 8 years ago
e718606
Make magjed@ owner of webrtc/api/android/
by magjed
· 8 years ago
64d6ff7
In VoiceEngine, the settings for APM are applied in such a way that
by peah
· 8 years ago
40217c3
Initial rate allocation should not use fps = 0
by sprang
· 8 years ago
57c1ad3
Don't declare function arguments of array type
by kwiberg
· 8 years ago
cc7bf88
Revert of Roll chromium_revision 5e821a778b..80ff2be807 (432715:433495) (patchset #1 id:1 of https://codereview.webrtc.org/2517933002/ )
by kjellander
· 8 years ago
6280960
Correctly pass drawn frame size when layout aspect ratio is used in EglRenderer.
by sakal
· 8 years ago
96c1587
RtpPacket::payload() return rtc::ArrayView instead of raw pointer
by danilchap
· 8 years ago
fe09560
Roll chromium_revision 5e821a778b..80ff2be807 (432715:433495)
by buildbot
· 8 years ago
f880285
iOS: Cleanup buildbot JSON files + bump iOS version to 10.0
by kjellander
· 8 years ago
3898944
Remove unused files linux.cc/.h and linuxfdwalk.c/.h.
by solenberg
· 8 years ago
2184155
Add more logging in ScreenCapturerIntegrationTest
by zijiehe
· 8 years ago
ed9dccf
Revert of Remove unused HttpClient class. (patchset #1 id:1 of https://codereview.webrtc.org/2511883005/ )
by honghaiz
· 8 years ago
4a698f6
Remove unused HttpClient class.
by solenberg
· 8 years ago
01af3a3
Remove unused dbus.cc/.h and related things.
by solenberg
· 8 years ago
90c024f
Move FirewallSocketServer to test code.
by nisse
· 8 years ago
00f2ee0
Changed the way we find the ProjectRootPath.
by ehmaldonado
· 8 years ago
dedaf1c
Modify audio_processing_unittest to use ResourcePath instead of ProjectRootPath.
by ehmaldonado
· 8 years ago
bbc747c
Delete WindowPicker class and subclasses.
by nisse
· 8 years ago
76b3049
Changed the interface AudioMixer::RemoveSource to have a void return type.
by aleloi
· 8 years ago
a28780e
Introduce ArrayView::subview function to return portion of the original view
by danilchap
· 8 years ago
509e4fe
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
by magjed
· 8 years ago
d7ac0a9
Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ )
by magjed
· 8 years ago
a82395b
Move smoothing filter to common audio.
by michaelt
· 8 years ago
610c454
Add Datachannel support to Android AppRTCMobile
by hekra01
· 8 years ago
1acfbd2
Expose RtpCodecParameters to VoiceMediaInfo stats.
by hbos
· 8 years ago
7b9feee
Fix PayloadRouter::OnEncodedImage() to handle errors properly.
by sergeyu
· 8 years ago
81c3a03
Added a callback function OnAddTrack to PeerConnectionObserver
by zhihuang
· 8 years ago
5b93db2
iOS: Add AudioSendSideBwe field trial.
by tkchin
· 8 years ago
eacbaea
Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ )
by magjed
· 8 years ago
0d0d753
Revert of Split out target rtc_media_base from rtc_media (patchset #3 id:40001 of https://codereview.webrtc.org/2471573003/ )
by magjed
· 8 years ago
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