Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
4b3c0d6f34c379c2a06f654b62403876b20180c6
4b3c0d6
Use WebRTC API to convert byteorder in srtpfilter.
by Jiayang Liu
· 9 years ago
4825356
RTCDataChannel: Unregister data channel observer on dealloc.
by Zeke Chin
· 9 years ago
379069f
VideoRenderCallback::RenderFrame: Make I420VideoFrame& ref const.
by Magnus Jedvert
· 9 years ago
0828a0c
Revert "Avoid critsect for protection- and qm setting callbacks in VideoSender."
by mflodman
· 9 years ago
23914fe
Reject RTP one-byte extension ID 0.
by Peter Boström
· 9 years ago
903c0f2
Avoid critsect for protection- and qm setting callbacks in VideoSender.
by mflodman
· 9 years ago
738a5b4
Remove old suppression for ProcessThreadImpl.
by Tommi
· 9 years ago
bc46bf2
common_audio: Explicit cast in WebRtcSpl_NormW16 on ARM
by Bjorn Volcker
· 9 years ago
0194d32
Add WebRtcAudioManager to peerconnection_jar library
by Alex Glaznev
· 9 years ago
65f74a1
Revert "Suppress data races in libjingle_peerconnection_unittest"
by Tommi
· 9 years ago
2c9c83d
Remove non-functional asynchronous resampling mode.
by Andrew MacDonald
· 9 years ago
45c6449
Introduce CodecManager and move code from AudioCodingModuleImpl
by Henrik Lundin
· 9 years ago
f7b9cf5
Suppress "EndToEndTest::ReceivedFecPacketsNotNacked" on Asan, Tsan
by Minyue Li
· 9 years ago
842a4a6
Add locks to Start(), Stop() methods in ProcessThread.
by Tommi
· 9 years ago
22e209d
Introduce AudioCodingModuleImpl::current_encoder_
by Henrik Lundin
· 9 years ago
582f80e
Clamp decoder sample rate to 32000 in iSAC
by Henrik Lundin
· 9 years ago
1ecfd55
videoadapter_unittest.cc: Revert removal of '#if defined(HAVE_WEBRTC_VIDEO)'
by Magnus Jedvert
· 9 years ago
451b614
Fix gyp path for bwe simulator include.
by Stefan Holmer
· 9 years ago
8e9c67e
Suppress data races in libjingle_peerconnection_unittest
by Henrik Kjellander
· 9 years ago
9f52448
Roll chromium_revision 4d63ee8..719b839 (322012:322539)
by Henrik Kjellander
· 9 years ago
6b3ccfc
GN: Cleanup no longer needed libvpx config.
by Henrik Kjellander
· 9 years ago
819011c
Additional suppression for TSan deadlock detection
by Henrik Kjellander
· 9 years ago
dfd53fe
Raise streams for SetMaxSendBitrates above 2000k.
by Peter Boström
· 9 years ago
53eda3d
Add tests for r8811.
by Peter Boström
· 9 years ago
b3fc48b
Update the notice about the slow Chromium sync.
by Henrik Kjellander
· 9 years ago
1d36003
Suppress TSan errors triggered when deadlock detection is enabled.
by Henrik Kjellander
· 9 years ago
9ff73f5
Final minor fix in WebRtcAudioManager
by henrika
· 9 years ago
424694c
audio_processing/agc: Put entire method set_output_will_be_muted() under lock
by Bjorn Volcker
· 9 years ago
75a0255
Handle borked Android cameras gracefully.
by Per
· 9 years ago
8324b52
Adding playout volume control to WebRtcAudioTrack.java.
by henrika
· 9 years ago
8ed6a4b
Remove unused non-standard capture stats.
by Peter Boström
· 9 years ago
3954e1d
Remove unused implementations in cricket::VideoFrame
by Magnus Jedvert
· 9 years ago
7100dcd
Adding "usedtx" as Opus codec parameter.
by Minyue Li
· 9 years ago
bef8d2d
Add a lock to NSSContext to fix data race
by Jiayang Liu
· 9 years ago
b8cfa68
Update speed setting in VP9.
by Marco
· 9 years ago
74d9ed7
Report send codec name in GetStats().
by Peter Boström
· 9 years ago
d6f4c25
Reject streams reusing simulcast or RTX SSRCs.
by Peter Boström
· 9 years ago
a990784
AcmReceiver: index decoders by payload type instead of ACM codec ID
by Jelena Marusic
· 9 years ago
9b5f96e
Add some sanity CHECKs to webrtc::Call.
by Peter Boström
· 9 years ago
c79f7ed
Fix build error introduced by r8864.
by Stefan Holmer
· 9 years ago
e590416
Moving the pacer and the pacer thread to ChannelGroup.
by Stefan Holmer
· 9 years ago
5225dd8
If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size.
by Brave Yao
· 9 years ago
dfa3605
Reparent Nonlinear beamformer under beamforming interface.
by Michael Graczyk
· 9 years ago
bf395c1
Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android
by Bjorn Volcker
· 9 years ago
caae5d4
Bye request should use POST not GET
by Chuck Hays
· 9 years ago
190c3ca
Register sample rate of Audio RED in RTPPayloadRegistry.
by Minyue Li
· 9 years ago
79064e5
Fix crash on decode found by fuzz tester.
by Stefan Holmer
· 9 years ago
3fbf99c
Refactor common_audio/vad: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT
by Bjorn Volcker
· 9 years ago
855acf7
Remove video from WebRTC Android example.
by Per
· 9 years ago
d4362cd
Reject StreamParams with RTX SSRCs not in ssrcs.
by Peter Boström
· 9 years ago
a49f515
Roll chromium_revision da9a1c0..4d63ee8 (321718:322012)
by Henrik Kjellander
· 9 years ago
1ccd8b4
Refactor common_audio/signal_processing: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT
by Bjorn Volcker
· 9 years ago
245989b
Address comments from cr 43769004.
by Tommi
· 9 years ago
0e209b0
Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/.
by Donald Curtis
· 9 years ago
e61c64d
Delete NullVideoRenderer
by Magnus Jedvert
· 9 years ago
07a4ba5
Simulcast settings for 1080p. Using same bit rates for all 3 modes since only one is used in reality, and the plan is to unify them.
by Niklas Enbom
· 9 years ago
ac27e20
Delete VideoAdapter::AdaptFrame
by Magnus Jedvert
· 9 years ago
45636ec
Post Git switch: Update codereview.settings and remove drover.properties
by Henrik Kjellander
· 9 years ago
68a5418
Enable PENDING_REF_PREFIX in codereview.settings.
by Henrik Kjellander
· 9 years ago
4d14592
rtc::Buffer: Restore length method for backwards compatibility
by kwiberg@webrtc.org
· 9 years ago
deafa7b
Remove I420VideoFrame::SwapFrame
by magjed@webrtc.org
· 9 years ago
2d2a30c
Remove I420VideoFrame::CloneFrame
by magjed@webrtc.org
· 9 years ago
0b52ceb
Improve logging and add DCHECKs in codec database.
by pbos@webrtc.org
· 9 years ago
eebcab5
rtc::Buffer: Rename length to size, for conformance with the STL
by kwiberg@webrtc.org
· 9 years ago
e815290
Update README instructions for Android AppRTCDemo.
by glaznev@webrtc.org
· 9 years ago
a5f6fb5
Permit single-stream max bitrates above 2000k.
by pbos@webrtc.org
· 9 years ago
a197a5e
Update libsrtp includes in preparation of roll into Chromium.
by jiayl@webrtc.org
· 9 years ago
a3ffc56
Allow setting thread priorities in Chromium on all but linux platforms.
by tommi@webrtc.org
· 9 years ago
39fc1d3
Disable PeerConnectionClientTest.testLoopbackVp9
by henrik.lundin@webrtc.org
· 9 years ago
0b44b58
Limit disabling of PeerConnectionEndToEndTest.Call to Windows
by henrik.lundin@webrtc.org
· 9 years ago
64eb2ff
iOS library build script
by tkchin@webrtc.org
· 9 years ago
9509fbf
Split EventWrapper in twain.
by tommi@webrtc.org
· 9 years ago
82e8ae4
Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest
by henrik.lundin@webrtc.org
· 9 years ago
2b4ce3a
Convert webrtc/video/ abort/assert to CHECK/DCHECK.
by pbos@webrtc.org
· 9 years ago
41d2bef
Limit RED audio payload to narrow band.
by minyue@webrtc.org
· 9 years ago
1596a4f
Temporarily disable SetPriority when building with Chromium.
by tommi@webrtc.org
· 9 years ago
d4e7d49
Scaler: Recycle allocations using buffer pool.
by magjed@webrtc.org
· 9 years ago
09b6ff9
Disable PLC for iSAC
by henrik.lundin@webrtc.org
· 9 years ago
ee0c5af
Remove unused version.py script.
by kjellander@webrtc.org
· 9 years ago
aa0bbab
Fix build failure
by jmarusic@webrtc.org
· 9 years ago
a4bef3e
AcmReceiver: use std::map instead of an array to keep the list of decoders
by jmarusic@webrtc.org
· 9 years ago
3335a4f
Prevent asserting on unset start bitrate.
by pbos@webrtc.org
· 9 years ago
50ed0d9
Roll chromium_revision 6311617..da9a1c0 (321517:321718)
by kjellander@webrtc.org
· 9 years ago
e5e92bd
Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix)
by kjellander@webrtc.org
· 9 years ago
cfde27e
Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows.
by kjellander@webrtc.org
· 9 years ago
38492c5
Re-land 8810 "- Add a SetPriority method to ThreadWr..."
by tommi@webrtc.org
· 9 years ago
90a1cb4
Revert 8810 "- Add a SetPriority method to ThreadWrapper"
by tommi@webrtc.org
· 9 years ago
b789f62
Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..."
by tommi@webrtc.org
· 9 years ago
0c34001
Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine."
by tommi@webrtc.org
· 9 years ago
346a64b
Mac would force bluetooth playout working with 8kHz/1ch if capturing/rendering shares the same device, e.g. changing from 44.1kHz/2ch as default.
by braveyao@webrtc.org
· 9 years ago
4553941
Document the 'int' return value of Resampler methods.
by wtc@chromium.org
· 9 years ago
3200a64
Minor fix for MIPS Android build.
by andrew@webrtc.org
· 9 years ago
4ddc938
Support VP8 hardware encoding and decoding on IA devices.
by glaznev@webrtc.org
· 9 years ago
b9557a9
Fix code to handle crashes for non-VP8.
by pbos@webrtc.org
· 9 years ago
b6817d7
- Add a SetPriority method to ThreadWrapper
by tommi@webrtc.org
· 9 years ago
66df3cf
Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
by pbos@webrtc.org
· 9 years ago
8296ec5
Fix heap-use-after-free in WebRtcVideoEngine2.
by pbos@webrtc.org
· 9 years ago
a3209a2
Release buffer pool in Vp8DecoderImpl::Release().
by pbos@webrtc.org
· 9 years ago
8904290
Make screenshare target bitrate experiment always on
by pbos@webrtc.org
· 9 years ago
d9c5024
Roll chromium_revision bd49b12..6311617 (320783:321517)
by kjellander@webrtc.org
· 9 years ago
Next »