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fp2-dev
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platform
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chromium_org
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third_party
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webrtc
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cd117d2ff3b413abca9a84500734a0dff5e22c14
cd117d2
Integrate fake_network_pipe into direct_transport.
by stefan@webrtc.org
· 11 years ago
0d8474d
Remove metrics_unittests
by kjellander@webrtc.org
· 11 years ago
ef1f6c3
Remove media_file from VideoEngine dependencies.
by pbos@webrtc.org
· 11 years ago
2a4595a
cpplint cleaning new API and its implementation files.
by mflodman@webrtc.org
· 11 years ago
b409d78
Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand().
by mflodman@webrtc.org
· 11 years ago
f22f12a
Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss.
by mflodman@webrtc.org
· 11 years ago
cc407fd
Make MouseCursor mutable
by sergeyu@chromium.org
· 11 years ago
32a0f69
audio_processing_unittest: unbreak clang compilation.
by fischman@webrtc.org
· 11 years ago
6b89cba
JNI Audio: remove dead members.
by fischman@webrtc.org
· 11 years ago
32705ce
Revert "Make MouseCursor mutable"
by sergeyu@chromium.org
· 11 years ago
0c7efa2
Make MouseCursor mutable
by sergeyu@chromium.org
· 11 years ago
4db3691
Stop transport in test SuspendBelowMinBitrate.
by pbos@webrtc.org
· 11 years ago
6f43aa7
Added method for getting default module state and protect agains a
by mflodman@webrtc.org
· 11 years ago
620d9e5
Modify video_render/ to allow a single old frame.
by pbos@webrtc.org
· 11 years ago
4494516
Delete capturers after destroying streams in test.
by pbos@webrtc.org
· 11 years ago
f3aed2f
Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..."
by asapersson@webrtc.org
· 11 years ago
b06cca3
Simplification of histogram normalization in delay estimator.
by bjornv@webrtc.org
· 11 years ago
39139dc
Revert r5294 to re-roll r5293.
by pbos@webrtc.org
· 11 years ago
2ec8a62
Adds robust validation functionality to the delay estimator
by bjornv@webrtc.org
· 11 years ago
beb643b
Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver
by sprang@webrtc.org
· 11 years ago
0af1d21
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
by turaj@webrtc.org
· 11 years ago
ee867fa
Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
by solenberg@webrtc.org
· 11 years ago
b8dc2e2
Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing.
by turaj@webrtc.org
· 11 years ago
f34e39b
Measure pacer queue size based on when packets are inserted rather than captured.
by stefan@webrtc.org
· 11 years ago
b50a841
Fix jitter buffer delay estimate.
by turaj@webrtc.org
· 11 years ago
7f0519e
Update talk to 58174641 together with http://review.webrtc.org/4319005/.
by wu@webrtc.org
· 11 years ago
ab6ccbc
Adding REMB to receive stream configuration, the send side will always
by mflodman@webrtc.org
· 11 years ago
9b3321f
Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
by asapersson@webrtc.org
· 11 years ago
9d9f138
Merge metrics_unittests into video_engine_tests.
by pbos@webrtc.org
· 11 years ago
d1dd1d2
Move realtime tests to webrtc_perf_tests.
by pbos@webrtc.org
· 11 years ago
0e4512b
Callback for send bitrate estimates - new roll
by sprang@webrtc.org
· 11 years ago
e4d538a
Make sure channels in the same call are in the same channel group.
by mflodman@webrtc.org
· 11 years ago
e6dc4ff
Making RemoteRateControl::min_configured_bit_rate_ configurable
by henrik.lundin@webrtc.org
· 11 years ago
3a4fc4b
Update talk to 58127566 together with
by wu@webrtc.org
· 11 years ago
b9cf1de
ACM 2 compatibility with ACM 1.
by turaj@webrtc.org
· 11 years ago
9b3d2bf
Revert 5274 "Update talk to 58113193 together with https://webrt..."
by wu@webrtc.org
· 11 years ago
cde78d6
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
by wu@webrtc.org
· 11 years ago
a4670a1
Complete rewrite of demo application.
by henrike@webrtc.org
· 11 years ago
0ceb51f
Remove overloaded CpuOveruseMeasure function.
by asapersson@webrtc.org
· 11 years ago
2f70422
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
by henrike@webrtc.org
· 11 years ago
f49d16c
Fix common_video_unittests in apk_tests.gyp.
by pbos@webrtc.org
· 11 years ago
7123a80
Add SwapFrame() to VideoSendStreamInput.
by pbos@webrtc.org
· 11 years ago
66e84b0
Revert 5259 "Callback for send bitrate estimates"
by sprang@webrtc.org
· 11 years ago
894dab9
Roll chromium_revision 232627:238260
by kjellander@webrtc.org
· 11 years ago
f1d22d4
Callback for send bitrate estimates
by sprang@webrtc.org
· 11 years ago
72b0d40
Removed unnecessary Pulse init from VoE startup.
by fischman@webrtc.org
· 11 years ago
e8ca064
Correctly define OVERRIDE when building with g++ 4.7 and C++11 support
by andrew@webrtc.org
· 11 years ago
090f37f
Improve VideoSendStreamTest::MaxPacketSize
by sprang@webrtc.org
· 11 years ago
ba8b32c
Change uses of the obsolete armv7 setting to arm_version==7.
by kjellander@webrtc.org
· 11 years ago
934be30
Fix compilation errors on Fedora 20.
by fischman@webrtc.org
· 11 years ago
4adc7ad
Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0.
by andrew@webrtc.org
· 11 years ago
e681a01
Add shape in DesktopFrame.
by sergeyu@chromium.org
· 11 years ago
e8dd108
Add new method to MockAudioProcessing.
by andrew@webrtc.org
· 11 years ago
e4d591a
Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..."
by andrew@webrtc.org
· 11 years ago
c8bd975
Allow opening an AEC dump from an existing file handle.
by henrikg@webrtc.org
· 11 years ago
9e40eba
Stop video capturers in multi-stream test.
by pbos@webrtc.org
· 11 years ago
5b23ce6
Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks
by sprang@webrtc.org
· 11 years ago
ed8c496
Fraction lost statistics not being reported
by sprang@webrtc.org
· 11 years ago
6dccf13
Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.
by sprang@webrtc.org
· 11 years ago
2de68d6
Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
by asapersson@webrtc.org
· 11 years ago
cf5c552
Add callbacks for send channel rtp statistics
by sprang@webrtc.org
· 11 years ago
f0d9b20
Remove CallTest dependency on voice_engine/test/.
by pbos@webrtc.org
· 11 years ago
8db148e
Add API to query video engine for the send-side delay.
by stefan@webrtc.org
· 11 years ago
adc238a
Fixing the android build
by henrik.lundin@webrtc.org
· 11 years ago
3d70641
Move implementation files out of the webrtc/ root.
by pbos@webrtc.org
· 11 years ago
b669e60
Remove default implementations for SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
3bcea52
Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics.
by stefan@webrtc.org
· 11 years ago
8911937
Add callbacks for send channel rtcp statistics
by sprang@webrtc.org
· 11 years ago
d90bebb
Make RTPSender::SendPadData public.
by stefan@webrtc.org
· 11 years ago
991d58c
Remove unused ThreadData struct.
by andrew@webrtc.org
· 11 years ago
5459e0b
Remove the long disabled WEBRTC_SVNREVISION define.
by andrew@webrtc.org
· 11 years ago
382cfdd
Removing DropDeltaAfterKey functionality which is unused.
by andresp@webrtc.org
· 11 years ago
9435a17
Add send frame rate statistics callback
by sprang@webrtc.org
· 11 years ago
f2c136b
Added a delay measurement, measures the time between an incoming captured frame until the frame is being processed. Measures the delay per second.
by asapersson@webrtc.org
· 11 years ago
0cb8020
Fixes a crash in fullstack tests introduced with r5209.
by stefan@webrtc.org
· 11 years ago
29a9669
Small fixes to plot_neteq_delay.m
by henrik.lundin@webrtc.org
· 11 years ago
da3ae7c
Adds support for sending redundant payloads over RTX.
by stefan@webrtc.org
· 11 years ago
c101a27
Fix a typo in neteq.gypi
by henrik.lundin@webrtc.org
· 11 years ago
c749348
Compile-out functions only used by the bit-exact test.
by andrew@webrtc.org
· 11 years ago
b1f4a72
Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close).
by fischman@webrtc.org
· 11 years ago
adaa8b5
Add baseline generation/verification to BWE test framework.
by solenberg@webrtc.org
· 11 years ago
4ed6832
Utility class for reading/writing network-byte-ordered integers.
by sprang@webrtc.org
· 11 years ago
397aae0
Change BitrateStats to more generalized RateStatistics
by sprang@webrtc.org
· 11 years ago
309b2c8
Set local SSRC for VideoReceiveStream.
by pbos@webrtc.org
· 11 years ago
f91a14f
Do not use recursive calling in NetEq test tools
by henrik.lundin@webrtc.org
· 11 years ago
02817f8
Fixing NetEq tests for new Opus version
by tina.legrand@webrtc.org
· 11 years ago
169a27a
Disable check for all sent SSRCs being valid.
by pbos@webrtc.org
· 11 years ago
758ef4c
This CL adds an API to enable robust validation of delay estimates.
by bjornv@webrtc.org
· 11 years ago
c8918cb
Fixes a crash in the pacer where it fails to find a normal prio packet if there are no high prio packets, given that the queue has grown too large.
by stefan@webrtc.org
· 11 years ago
0e2571d
Lock frame in ViECapturer::IncomingFrameI420.
by pbos@webrtc.org
· 11 years ago
9105cbd
Set up SSRCs correctly after switching codec.
by pbos@webrtc.org
· 11 years ago
b8b2a23
Recommit CL5184
by bjornv@webrtc.org
· 11 years ago
151cd25
Refactor Remote Estimators Test into a more reusable form.
by solenberg@webrtc.org
· 11 years ago
66d634f
Revert 5184 "Small refactoring change in delay_estimator."
by bjornv@webrtc.org
· 11 years ago
2c75d4e
Small refactoring change in delay_estimator.
by bjornv@webrtc.org
· 11 years ago
801822c
Ensure that no packet stays in the pacer queue for longer than 2 seconds.
by stefan@webrtc.org
· 11 years ago
8f9da30
Create default implementation to fix issue in libjingle
by sprang@webrtc.org
· 11 years ago
4a9843f
Implement and test EncodedImageCallback in new ViE API.
by sprang@webrtc.org
· 11 years ago
2622be1
Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api.
by asapersson@webrtc.org
· 11 years ago
58b912b
Remove const in vie_rtp_rtcp, where there is conflict with
by sprang@webrtc.org
· 11 years ago
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