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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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444682acf9804c5fcbddaded9e900ba3cc6921fc
444682a
Remove frame time scheduing in IncomingVideoStream
by qiangchen
· 9 years ago
953eabc
Revert "GetDefaultLocalAddress should return false when the address is invalid"
by Guo-wei Shieh
· 9 years ago
67c6df6
GetDefaultLocalAddress should return false when the address is invalid
by Guo-wei Shieh
· 9 years ago
7d842d6
Move thread_ conditional back under defines.
by Peter Boström
· 9 years ago
c661213
Skip setting thread priorities in NaCl.
by Peter Boström
· 9 years ago
b251472
Add JNI interface for functions to start and stop recording AEC dumps and RTC event logs.
by ivoc
· 9 years ago
c3ddb3e
Improve documentation for ArrayView
by kwiberg
· 9 years ago
4c5eea3
Android SurfaceViewRenderer: Don't rely on widthSpec/heightSpec after onMeasure() returns
by Magnus Jedvert
· 9 years ago
b7a8829
Remove duplicated headers after updating downstream code.
by kjellander
· 9 years ago
302c978
Work around data race in TransmitMixer.
by solenberg
· 9 years ago
7baf79f
Temporary remove spamming MediaDecoder log
by perkj
· 9 years ago
92f8dbd
Remove VIDEOCODEC_* from engine_configurations.h.
by Peter Boström
· 9 years ago
4f2152e
Android SurfaceViewRenderer: Make sure not to call eglCreateSurface() twice
by Magnus Jedvert
· 9 years ago
97c821d
Inline ConvertToSystemPriority.
by Peter Boström
· 9 years ago
9237559
Add SurfaceTextureHelper.disconnect(Handler handler) method
by perkj
· 9 years ago
d480153
Add option to capture to texture in AppRTCDemo for Android.
by Per
· 9 years ago
978244e
Adding a bunch of Agora IO team members to the watch lists
by yujie.mao
· 9 years ago
d860523
First part of the preparatory work before the actual work for solving the ducking problem starts.
by peah
· 9 years ago
70bed7d
GN: Fix iOS error in audio_device and rtc_base
by kjellander
· 9 years ago
b5cb19b
Fixing direction attribute in answer for non-RTP protocols.
by deadbeef
· 9 years ago
05816eb
Fix target_arch for ios devices
by wr.wllm
· 9 years ago
12411ef
Move ThreadWrapper to ProcessThread in base.
by pbos
· 9 years ago
255d6f6
Test case for CL 1437933002.
by guoweis
· 9 years ago
9c80bbe
Roll chromium_revision e038f1d..b1d79c3 (361088:361146)
by kjellander
· 9 years ago
1aa6efe
Android ThreadUtils: Make the class public for access outside org.webrtc
by Magnus Jedvert
· 9 years ago
057fb89
Add new method AcmReceiver::last_packet_sample_rate_hz()
by henrik.lundin
· 9 years ago
74e35f1
Remove the special case for std::vector in rtc::ArrayView
by kwiberg
· 9 years ago
d89814b
NetEq: Add new method last_output_sample_rate_hz
by henrik.lundin
· 9 years ago
dfafd12
Remove ThreadWrapper::GetThreadId. The method just calls rtc::CurrentThreadId(), which also has a more descriptive name.
by Tommi
· 9 years ago
62e9bda
Implement fuzzing of VP9 depacketization.
by Peter Boström
· 9 years ago
ee37de3
Add screenshare perf tests with lossy links
by sprang
· 9 years ago
1379f1f
Extract the parameters for the encoder stack from the CodecManager
by kwiberg
· 9 years ago
30a5e56
Roll chromium_revision 3796a7a..e038f1d (361065:361088)
by kjellander
· 9 years ago
f0a431a
Exclude EndToEndTest.ReceivesTransportFeedback and TransportFeedbackNotConfigured from DrMemory.
by Stefan Holmer
· 9 years ago
db81ffd
Request keyframe if too many packets are missing and NACK is disabled.
by jbauch
· 9 years ago
fa8ae9a
Remove <iostream> include from file_audio_device.cc
by kjellander@webrtc.org
· 9 years ago
8becec3
talk: remove deprecated *processor.h files
by tfarina
· 9 years ago
87d5845
Fix androidmediadecoder_jni TS logging.
by perkj
· 9 years ago
c3c4cdb
Add Android x86 and x64 trybots to CQ.
by kjellander@webrtc.org
· 9 years ago
d5674c3
Roll chromium_revision c29f20c..3796a7a (361043:361065)
by kjellander
· 9 years ago
50c5136
RTCP Bye packet moved to own file
by danilchap
· 9 years ago
c982913
Roll chromium_revision 6018759..c29f20c (361030:361043)
by kjellander
· 9 years ago
485b5b0
Roll chromium_revision 4df2d47..6018759 (361029:361030)
by kjellander
· 9 years ago
82581a0
Roll chromium_revision 3966d2c..4df2d47 (361020:361029)
by kjellander
· 9 years ago
b4a29d9
Roll chromium_revision b854092..3966d2c (360794:361020)
by kjellander
· 9 years ago
13f6b8f
Increase transport feedback frequency to 20 Hz.
by stefan
· 9 years ago
43edf0f
Require negotiation to send transport cc feedback over RTCP.
by stefan
· 9 years ago
bd13838
Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack.
by solenberg
· 9 years ago
672304a
NetEq: Remove overly verbose logging
by henrik.lundin
· 9 years ago
5def7b9
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
by deadbeef
· 9 years ago
7add058
Move some receive stream configuration into webrtc::AudioReceiveStream.
by solenberg
· 9 years ago
6834fa1
Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
by deadbeef
· 9 years ago
0a43fef
Allow pacer to boost bitrate in order to meet time constraints.
by sprang
· 9 years ago
34911ad
Improved error handling in iOS ADM to avoid race during init
by henrika
· 9 years ago
76a31ca
Avoids hitting DCHECK in OpenSL ES player
by henrika
· 9 years ago
1afae74
Roll chromium_revision 5c83f4e..b854092 (360728:360794)
by kjellander
· 9 years ago
30e9182
This cl add support to encode from textures to MediaCodecVideoEncoder.
by perkj
· 9 years ago
5663b4f
iOS: Set enable_protobuf=1 by default.
by kjellander@webrtc.org
· 9 years ago
7e63ef0
Allow default audio receive channel to receive on any unsignalled SSRC.
by solenberg
· 9 years ago
b0ad43b
Add aecdump support to audioproc_f
by aluebs
· 9 years ago
ceb450b
Roll chromium_revision c8eec9a..5c83f4e (360565:360728)
by kjellander
· 9 years ago
17c0aff
Enable VP9 HW decoder on Exynos chips.
by Alex Glaznev
· 9 years ago
7593aad
Re-enable mistakenly disabled PEM tests. Misc cleanup and alignment fixes.
by torbjorng
· 9 years ago
7755e20
Chrome has now been updated.
by perkj
· 9 years ago
726b1f7
Removed dummy "mediastreamsignaling.h"
by perkj
· 9 years ago
191c1f9
Disable all JsepPeerConnectionP2PTestClient tests on Mac due to flakiness on Mac Debug bots.
by ivoc
· 9 years ago
12e21a0
Remove dead code (we no longer support SILK)
by kwiberg
· 9 years ago
ef45323
Android: Make classes non-final
by Magnus Jedvert
· 9 years ago
062e14e
Roll chromium_revision bb7899a..c8eec9a (360504:360565)
by kjellander
· 9 years ago
f399f21
Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 on linux due to flakiness on the Linux64 Debug bot.
by ivoc
· 9 years ago
f22695c
Remove build_with_libjingle and exclude failing iOS tests from 'All' target.
by kjellander@webrtc.org
· 9 years ago
1503867
Disabled several JsepPeerConnectionP2PTestClient tests on Mac, due to flakiness on Debug Mac trybots.
by ivoc
· 9 years ago
e488a0d
Fix DTLS packet boundary handling in SSLStreamAdapterTests.
by jbauch
· 9 years ago
87097a8
Roll chromium_revision ed2e3fb..bb7899a (360379:360504)
by kjellander
· 9 years ago
b6755ab
Revert of Adding thread timeout for audio recorer thread in Java (patchset #2 id:20001 of https://codereview.webrtc.org/1444313002/ )
by henrika
· 9 years ago
488e75f
Patchset 1 yet again relands without modification https://codereview.webrtc.org/1422963003/
by Per
· 9 years ago
0969398
Revert of Remove android_rel from CQ since all of its machines are offline. (patchset #1 id:1 of https://codereview.webrtc.org/1459083002/ )
by kjellander
· 9 years ago
392d0c2
Remove android_rel from CQ since all of its machines are offline.
by Henrik Kjellander
· 9 years ago
521ed7b
Reland Convert internal representation of Srtp cryptos from string to int
by Guo-wei Shieh
· 9 years ago
318166b
Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
by guoweis
· 9 years ago
2764e10
Convert internal representation of Srtp cryptos from string to int.
by guoweis
· 9 years ago
3c652b6
modules/audio_coding: Remove some codec include dirs
by kjellander@webrtc.org
· 9 years ago
b7ce964
modules/video_coding/utility: Remove include
by kjellander@webrtc.org
· 9 years ago
1b20d81
Roll chromium_revision 64f2817..ed2e3fb (360275:360379)
by kjellander
· 9 years ago
0f59a88
modules/video_processing: refactor interface->include + more.
by Henrik Kjellander
· 9 years ago
ed7d6ec
WebRTC: Add compability header for video_coding refactoring.
by Henrik Kjellander
· 9 years ago
ad948c4
Preliminary support of VP9 HW encoder on Android.
by Alex Glaznev
· 9 years ago
2557b86
modules/video_coding refactorings
by Henrik Kjellander
· 9 years ago
4dd7a65
Temporarily disable VERIFY while bug is investigated.
by phoglund
· 9 years ago
223692a
Remove dead code
by kwiberg
· 9 years ago
e1a27d4
Move CNG/RED payload type extraction to Rent-A-Codec
by kwiberg
· 9 years ago
49a6c99
Disables BitrateEstimatorTest.SwitchesToASTThenBackToTOFForVideo on win_drmemory_full due to flakiness.
by ivoc
· 9 years ago
2446e5a
Fixed the render queue item size preallocation and verification, moved constant declarations, removed redundant queue allocation
by peah
· 9 years ago
0219c9b
rtcp::App moved into own file and got Parse function
by danilchap
· 9 years ago
2aff615
Remove spammy logging of RTCP delivery failures.
by Peter Boström
· 9 years ago
f70568c
So long and thanks for all the code reviews!
by andrew
· 9 years ago
cb50c96
Set temporal up switch bit to false for flexible mode (one temporal layer is configured currently).
by asapersson
· 9 years ago
aa45843
Roll chromium_revision a6d9f7f..64f2817 (360123:360275)
by kjellander
· 9 years ago
310b093
Fix active tcp port to 9
by Guo-wei Shieh
· 9 years ago
2935e01
Several Tick counter improvements try #2."
by thaloun
· 9 years ago
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