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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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28e9b66210667790f4ce46f5b0bfca1ebd32b212
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video
60f1422
Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.
by wu@webrtc.org
· 10 years ago
5e44f56
* Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter.
by wu@webrtc.org
· 10 years ago
c54ff69
Add thread annotations to Call API.
by pbos@webrtc.org
· 10 years ago
267637b
Disable flaky CaptureNtpTimeWithNetworkJitter.
by pbos@webrtc.org
· 10 years ago
c298835
Disabling flaky CanReceiveFec.
by pbos@webrtc.org
· 10 years ago
b991cd0
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
by wu@webrtc.org
· 10 years ago
dbebc39
Remove TraceCallback use from Call.
by pbos@webrtc.org
· 10 years ago
9d0f79f
Rename Start/Stop in Video{Send,Receive}Streams.
by pbos@webrtc.org
· 10 years ago
69b14d5
Let A/V sync test use default AudioCoding module
by henrik.lundin@webrtc.org
· 10 years ago
6b1114a
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
by fischman@webrtc.org
· 10 years ago
19ca463
Disable UsesTraceCallback
by pbos@webrtc.org
· 10 years ago
d8b4d0f
Implement FEC support in VideoReceiveStream.
by pbos@webrtc.org
· 10 years ago
4b0cd7f
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
by andresp@webrtc.org
· 10 years ago
1982636
Clean up traces and logs in RemoteBitrateEstimator.
by stefan@webrtc.org
· 10 years ago
6c57efd
Re-submit: rev5775
by andresp@webrtc.org
· 10 years ago
765ea72
Revert 5775 "Modify bitrate controller to update bitrate based o..."
by andrew@webrtc.org
· 10 years ago
0b11715
Change sprintf format string from %zu to %i
by henrik.lundin@webrtc.org
· 10 years ago
539bbde
Modify bitrate controller to update bitrate based on process call and not
by andresp@webrtc.org
· 10 years ago
5f804f8
VoE changes to allow forwarding of packets from VoE to ViE BWE.
by solenberg@webrtc.org
· 10 years ago
9b2b8ec
Add AIMD option to BWE API.
by stefan@webrtc.org
· 10 years ago
a183edc
Extend perf tests to perform rampup on single stream.
by andresp@webrtc.org
· 10 years ago
292e7f6
Disabling SendsSetSimulcastSsrcs.
by pbos@webrtc.org
· 10 years ago
16c3dcc
Disable flaky CanSwitchToUseAllSsrcs.
by pbos@webrtc.org
· 10 years ago
bef6e62
Simplify pacer interface.
by pbos@webrtc.org
· 10 years ago
f39df52
Remove internal codecs from VideoSendStream.
by pbos@webrtc.org
· 10 years ago
ebae8bb
Re-comitting r5711: "Fixing a flaky test in video_engine_tests"
by henrik.lundin@webrtc.org
· 10 years ago
8d3c410
Revert 5711 "Fixing a flaky test in video_engine_tests"
by turaj@webrtc.org
· 10 years ago
f9a6ab0
Fixing a flaky test in video_engine_tests
by henrik.lundin@webrtc.org
· 10 years ago
ca626eb
Refactor rampup tests:
by andresp@webrtc.org
· 10 years ago
3c00b1c
Stopping network threads before tearing down test
by henrik.lundin@webrtc.org
· 10 years ago
15cf717
Re-landing "Routing SuspendChange to VideoSendStream::Stats"
by henrik.lundin@webrtc.org
· 10 years ago
9420a1f
Implement minimum transmit bitrate.
by pbos@webrtc.org
· 10 years ago
41da329
Enable all RampUpTest.UpDownUp* tests
by henrik.lundin@webrtc.org
· 10 years ago
c53e587
Replace labs with std::abs.
by pbos@webrtc.org
· 10 years ago
af634a2
Remove platform-specific code from new-API tests.
by pbos@webrtc.org
· 10 years ago
f951dfc
Revert "Routing SuspendChange to VideoSendStream::Stats"
by henrik.lundin@webrtc.org
· 10 years ago
697cd78
Routing SuspendChange to VideoSendStream::Stats
by henrik.lundin@webrtc.org
· 10 years ago
f422ce1
Adding a link to issue
by henrik.lundin@webrtc.org
· 10 years ago
b5c0d2e
NetEq4: Changing the behavior of playout_timestamp_ update
by henrik.lundin@webrtc.org
· 10 years ago
4368a8f
Potential deadlock in VideoSendStreamTest::ProducesStats
by sprang@webrtc.org
· 10 years ago
c63f18d
Use DISABLE_ instead of commenting out tests
by henrik.lundin@webrtc.org
· 10 years ago
0bf5a2f
Adding a new ramp-up-down-up test
by henrik.lundin@webrtc.org
· 10 years ago
deb5d53
Fix compilation errors under clang 3.5.
by pbos@webrtc.org
· 10 years ago
3f3e951
Incorrect overhead calculation when using FEC + RTP extension headers.
by sprang@webrtc.org
· 10 years ago
f2c28a0
Make VideoReceiveStream::GetStats() const.
by sprang@webrtc.org
· 10 years ago
fa7c4c4
Wire up statistics in video receive stream of new API
by sprang@webrtc.org
· 10 years ago
4b1817f
Add configuration for cpu overuse detection to video send stream.
by asapersson@webrtc.org
· 10 years ago
fdb30d1
Fix race when deleting video receive streams in Call.
by solenberg@webrtc.org
· 11 years ago
48ac0da
Drop early packets when not sending in TransportAdapter.
by sprang@webrtc.org
· 11 years ago
ffd4269
Always initialize Trace in Call TraceDispatcher.
by pbos@webrtc.org
· 11 years ago
c766775
Wire up RTX in VideoReceiveStream.
by pbos@webrtc.org
· 11 years ago
bb55b6d
Set NACKed packet to -1 in TestNackRetransmission.
by pbos@webrtc.org
· 11 years ago
64339f0
Add configuration and test for extended RTCP reference time reports to new video api.
by asapersson@webrtc.org
· 11 years ago
df9f099
Permitting double start/stopping of streams.
by pbos@webrtc.org
· 11 years ago
c92ae91
Add another test case for AST/TOF switching.
by solenberg@webrtc.org
· 11 years ago
ca72300
Wire up statistics in video send stream of new video engine api
by sprang@webrtc.org
· 11 years ago
acc2e43
Add callbacks for receive channel RTCP statistics.
by sprang@webrtc.org
· 11 years ago
cd117d2
Integrate fake_network_pipe into direct_transport.
by stefan@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
4db3691
Stop transport in test SuspendBelowMinBitrate.
by pbos@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
39139dc
Revert r5294 to re-roll r5293.
by pbos@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
ab6ccbc
Adding REMB to receive stream configuration, the send side will always
by mflodman@webrtc.org
· 11 years ago
d1dd1d2
Move realtime tests to webrtc_perf_tests.
by pbos@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
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
7123a80
Add SwapFrame() to VideoSendStreamInput.
by pbos@webrtc.org
· 11 years ago
090f37f
Improve VideoSendStreamTest::MaxPacketSize
by sprang@webrtc.org
· 11 years ago
e4d591a
Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..."
by andrew@webrtc.org
· 11 years ago
9e40eba
Stop video capturers in multi-stream test.
by pbos@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
f0d9b20
Remove CallTest dependency on voice_engine/test/.
by pbos@webrtc.org
· 11 years ago
3d70641
Move implementation files out of the webrtc/ root.
by pbos@webrtc.org
· 11 years ago
da3ae7c
Adds support for sending redundant payloads over RTX.
by stefan@webrtc.org
· 11 years ago
309b2c8
Set local SSRC for VideoReceiveStream.
by pbos@webrtc.org
· 11 years ago
169a27a
Disable check for all sent SSRCs being valid.
by pbos@webrtc.org
· 11 years ago
9105cbd
Set up SSRCs correctly after switching codec.
by pbos@webrtc.org
· 11 years ago
4a9843f
Implement and test EncodedImageCallback in new ViE API.
by sprang@webrtc.org
· 11 years ago
586becf
Add test for automatically disabling padding when no video is being captured.
by stefan@webrtc.org
· 11 years ago
d7d60c8
Connect pacer/padding to SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
f8486d0
Rename DestroyStream methods to include Video.
by pbos@webrtc.org
· 11 years ago
04bcc9d
Deliver I420VideoFrames from VideoRender module.
by pbos@webrtc.org
· 11 years ago
f3b4602
Rename newapi::Transport::SendRTP()->SendRtp().
by pbos@webrtc.org
· 11 years ago
60108c2
Rename RTP-extension constants.
by pbos@webrtc.org
· 11 years ago
48cc9dc
Rename video streams' start/stop methods.
by pbos@webrtc.org
· 11 years ago
162021c
Rename Call::Create{Receive,Send}Stream().
by pbos@webrtc.org
· 11 years ago
8fdf191
Rename AutoMute to SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
26a736f
Hook up audio/video sync to Call.
by stefan@webrtc.org
· 11 years ago
5eca0c7
Fix breakage after introducing new test.
by stefan@webrtc.org
· 11 years ago
f8c47a1
Improve Call tests for RTX.
by stefan@webrtc.org
· 11 years ago
8f2997c
Implement VideoSendStream::SetCodec().
by pbos@webrtc.org
· 11 years ago
9a1635a
Make video/ only depend on video_engine_core.
by pbos@webrtc.org
· 11 years ago
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