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gerrit-public.fairphone.software
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fp2-dev
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platform
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external
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chromium_org
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third_party
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webrtc
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29fce828c804e309f9bd0ae9e3e25312caf70722
29fce82
To use the channel_transport on the iOS platform, some #if directives are changed.
by sjlee@webrtc.org
· 11 years ago
e8eaed8
Call AllowCommandLineReparsing in unit tests.
by andrew@webrtc.org
· 11 years ago
0180fc4
Split video coding module unit tests into sender and receiver unit tests.
by andresp@webrtc.org
· 11 years ago
f952fce
Implement NACK over RTX for VideoSendStream.
by pbos@webrtc.org
· 11 years ago
3b6ab4a
Remove use of vcm->ResetDecoder from modules/utility.
by andresp@webrtc.org
· 11 years ago
91b0d23
Allocate float_buffer_ in the initializer list.
by andrew@webrtc.org
· 11 years ago
f458c43
Split VideoCodingModuleImpl into VideoSender and VideoReceiver.
by andresp@webrtc.org
· 11 years ago
e125ca7
Prepare to compile ACM1 and ACM2.
by turaj@webrtc.org
· 11 years ago
cda8e61
Implement DesktopRegion subtraction.
by sergeyu@chromium.org
· 11 years ago
564ba1e
Moving test-only code (stream_generator) out of vcm implemention.
by andresp@webrtc.org
· 11 years ago
1d8ceab
Fix win trybot errors due to r4729.
by andrew@webrtc.org
· 11 years ago
985848d
Fix crash in the window capturer on windows
by sergeyu@chromium.org
· 11 years ago
242b8a5
ACM2 integration with NetEq 4.
by turaj@webrtc.org
· 11 years ago
54164d5
Adding Ami to the video renderer and capturer modules.
by mallinath@webrtc.org
· 11 years ago
4d57e48
The video render module for iOS.
by fischman@webrtc.org
· 11 years ago
1963a68
This issue is related to https://chromereviews.googleplex.com/9908014/
by minyue@webrtc.org
· 11 years ago
a5b7b8c
Make PCM16 available in Chromium builds.
by andrew@webrtc.org
· 11 years ago
40bd492
Make the destructor of AudioCodingModule public.
by andrew@webrtc.org
· 11 years ago
93da8cb
Fix unsigned/signed comparison error due to r4729.
by andrew@webrtc.org
· 11 years ago
e45a8a8
Reduce frequency of high audio delay warning logs.
by andrew@webrtc.org
· 11 years ago
9d775a6
Removes function that is not used anywhere but somehow still causing library load issues on Android Release build.
by henrike@webrtc.org
· 11 years ago
e22b761
Implement 'abs-send-time' extension in VideoSendStream.
by pbos@webrtc.org
· 11 years ago
75e7cff
OpenSl: fixes crashes externally reported in issue 2361 and 2362.
by henrike@webrtc.org
· 11 years ago
811e4c9
Adding APIs. These APIs are not implemented yet, they are to help developement of ACM.
by turaj@webrtc.org
· 11 years ago
7efd262
Remove FrameForStorage:Follow up on r4688
by mikhal@webrtc.org
· 11 years ago
905cebd
Implement 'toffset' extension in VideoSendStream.
by pbos@webrtc.org
· 11 years ago
910520a
Reset jitter buffer and timing if frames are getting too much delay.
by stefan@webrtc.org
· 11 years ago
96da891
Remove repeated conditions key.
by andrew@webrtc.org
· 11 years ago
3965d1f
OpenSL (not default): Enables low latency audio on Android.
by henrike@webrtc.org
· 11 years ago
4c94668
Fix format string in video_quality_analysis.cc.
by pbos@webrtc.org
· 11 years ago
d1deeb6
Remove include_dirs from voice_engine.gyp.
by pbos@webrtc.org
· 11 years ago
af73083
Test that VideoSendStream responds to NACK.
by pbos@webrtc.org
· 11 years ago
b1b278e
Convert printing in video quality tests to Chromium's perf format.
by kjellander@webrtc.org
· 11 years ago
0a477d1
Lock RTPSender statistics.
by pbos@webrtc.org
· 11 years ago
4d1cb14
Split up EngineTests and RampupTests.
by pbos@webrtc.org
· 11 years ago
81c4d24
Remove redundant STR_CASE_CMP macro definitions.
by andrew@webrtc.org
· 11 years ago
98691c2
Updated WebRTC version to 3.41
by elham@webrtc.org
· 11 years ago
0313e5b
Lock use of _packetRequestCallback in VCM.
by pbos@webrtc.org
· 11 years ago
462460f
Remove include_dirs from video_engine_core.gypi.
by pbos@webrtc.org
· 11 years ago
38ba534
Break out RTCPSender dependency on ModuleRtpRtcpImpl.
by pbos@webrtc.org
· 11 years ago
fdc4352
Rename VideoCall to Call.
by pbos@webrtc.org
· 11 years ago
9c843fd
Re-enable tests for Remote Bitrate Estimator
by solenberg@webrtc.org
· 11 years ago
0ee03f9
ExternalVideoDecoder for new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
b8aa042
Handle empty RTP video packets agnostic to codec.
by pbos@webrtc.org
· 11 years ago
d44ec1c
Reduce cost of PushSincResampler::Resample().
by andrew@webrtc.org
· 11 years ago
252b16f
Clamp camera id to legal values.
by fischman@webrtc.org
· 11 years ago
5ee7139
Improving padding rules and breaking out bw allocation to ViEEncoder.
by stefan@webrtc.org
· 11 years ago
5632a64
Adds support for combining RTX and FEC/RED.
by stefan@webrtc.org
· 11 years ago
b49897c
Add temporal layer factory.
by andresp@webrtc.org
· 11 years ago
4a4d15b
Removing FrameForStorage
by mikhal@webrtc.org
· 11 years ago
1e88712
Make unittest log printouts opt-in with a --logs flag.
by andrew@webrtc.org
· 11 years ago
5a196e6
Pre-multiply images for MouseCursorShape.
by alexeypa@chromium.org
· 11 years ago
7ac916b
Recognize armv7 target_arch for ios support in webrtc common.gyp
by fischman@webrtc.org
· 11 years ago
744235e
Restore severity precondition to logging.h.
by andrew@webrtc.org
· 11 years ago
6da93db
Remove send and receive streams when destroyed.
by pbos@webrtc.org
· 11 years ago
cade3c3
Add clockdrift to RtpGenerator
by henrik.lundin@webrtc.org
· 11 years ago
882b499
Allow unknown flags in test_main.cc.
by pbos@webrtc.org
· 11 years ago
5b5cf3c
NetEq4: Small change to reduce allocs in AudioMultiVector
by henrik.lundin@webrtc.org
· 11 years ago
934ddca
Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter.
by mflodman@webrtc.org
· 11 years ago
9d1d4b1
Clean capture timestamp code.
by andresp@webrtc.org
· 11 years ago
9333ee7
Disable EngineTest.ReceivesPliAndRecoversWithNack.
by mflodman@webrtc.org
· 11 years ago
d7e6388
Protecting Bitrate to avoid data race found by tsan.
by mflodman@webrtc.org
· 11 years ago
b1af9a8
Revert 4671 "Enable SetInitialPlayoutDelay on Android."
by mflodman@webrtc.org
· 11 years ago
8f34f73
Enable SetInitialPlayoutDelay on Android.
by dwkang@webrtc.org
· 11 years ago
179fc03
Don't force cont' when enabling kWithErrors
by mikhal@webrtc.org
· 11 years ago
1a4a552
Removing some TODO's from libyuv
by mikhal@webrtc.org
· 11 years ago
29befe4
Removing non decodable count from session info: This value isn't used, and therfore can be removed. This is a step towards the refactor of the session info to use maps.
by mikhal@webrtc.org
· 11 years ago
0e84525
Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule.
by mflodman@webrtc.org
· 11 years ago
43ec357
Add FakeEncoder to VideoSendStream tests.
by pbos@webrtc.org
· 11 years ago
777e192
Correcting two nits in InputAudioFile
by henrik.lundin@webrtc.org
· 11 years ago
c88d905
Changed method name.
by mflodman@webrtc.org
· 11 years ago
81ef3b8
Renamed method.
by mflodman@webrtc.org
· 11 years ago
a911872
Function name change.
by mflodman@webrtc.org
· 11 years ago
b7f97fc
Fixing capture frame race in ViECapturer.
by mflodman@webrtc.org
· 11 years ago
efdafa9
Disable all LS_VERBOSE logging in NetEq4
by henrik.lundin@webrtc.org
· 11 years ago
0cb9df1
NetEq4: Make the algorithm buffer a member variable
by henrik.lundin@webrtc.org
· 11 years ago
ee42c34
Overuse detection based on capture-input jitter.
by pbos@webrtc.org
· 11 years ago
fd74f30
Removing JPEG as it is not used.
by mikhal@webrtc.org
· 11 years ago
7a776d2
Zero comfort noise for stereo insted of assertion.
by turaj@webrtc.org
· 11 years ago
e0c5f92
Reorder and add critical section to the public method NetEqImpl::PacketBufferStatistics().
by turaj@webrtc.org
· 11 years ago
b266bf3
Fix typo in InvertedDesktopFrame
by sergeyu@chromium.org
· 11 years ago
5566bbd
Fix fileutils.cc for tests running under Win memory tools.
by kjellander@webrtc.org
· 11 years ago
b384812
Disabling CondVarTest for TSan v2 (take 2)
by kjellander@webrtc.org
· 11 years ago
a8096d6
Update SSRC in RtpRtcp for audio channel so that it can have NTP values for further AV sync.
by dwkang@webrtc.org
· 11 years ago
5422312
update neteq 4 to facilitate NACK
by minyue@webrtc.org
· 11 years ago
d09d996
Fix metrics_unittests on Android.
by kjellander@webrtc.org
· 11 years ago
fd9b155
Add isolate configuration for Android for all tests.
by kjellander@webrtc.org
· 11 years ago
cb08bc2
Re-organizing ACM tests
by tina.legrand@webrtc.org
· 11 years ago
a3eb5f7
Revert r4562
by elham@webrtc.org
· 11 years ago
d3aa1cc
Fix image flipping for OpenGL-based screen capturer on Mac.
by sergeyu@chromium.org
· 11 years ago
a7a3eae
Enable ObjC build by default and reenable 64-bit mac libjingle build
by fischman@webrtc.org
· 11 years ago
142ff66
Updated WebRTC version to 3.40
by elham@webrtc.org
· 11 years ago
d6a0007
VCM:Accounting for bounds when inserting packets. We currently receive indicators to the first and last packets of the frame, but not have any sanity to verify that all packets are indeed within the bounds (when available). This cl attempts to fix that,
by mikhal@webrtc.org
· 11 years ago
e5b027b
Relanding 4597 - Don't force key frame when decoding with errors.
by mikhal@webrtc.org
· 11 years ago
cf6bc76
WindowCapturer implementation for Linux.
by sergeyu@chromium.org
· 11 years ago
3b68458
Disables RtpRtcpTest.CanTransmitExtraRtpPacketsWithoutError as it flakily breaks the waterfall. See http://chromegw.corp.google.com/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/99/steps/voe_auto_test/logs/stdio the cl triggering it was a no-change (disabled some other broken tests).
by henrike@webrtc.org
· 11 years ago
d8e92c9
Remove newapi:: namespace for typenames without overlap.
by pbos@webrtc.org
· 11 years ago
b0e1da0
Revert 4597 "Don't force key frame when decoding with errors"
by henrike@webrtc.org
· 11 years ago
f186b23
Implement window capturer for OS X.
by sergeyu@chromium.org
· 11 years ago
e9b9d24
Don't force key frame when decoding with errors
by mikhal@webrtc.org
· 11 years ago
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