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2014-03-21Adding operator== and != methods for CodecInst and VideoCodec structures.mallinath@webrtc.org
R=juberti@google.com, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10099005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5746 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20Add ability to configure cpu overuse options via an API.asapersson@webrtc.org
BUG=1577 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9299006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5736 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19Fixes RTX related bugs.stefan@webrtc.org
- An RTX packet with no payload should be dropped prior to parsing RTX header since it doesn't have an RTX header. This can for example happen when sending padding-only packets over the RTX stream. - The retransmit code path when the pacer is disabled doesn't properly update the abs-send-time and ts-offset header extensions. TEST=trybots R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9189004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5728 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19Simplify pacer interface.pbos@webrtc.org
New interface uses two bitrates (max/min). The pace multiplier is also removed from the interface and instead utilized outside. Min bitrate will be filled with padding if there's not enough media to transmit. Also fixes a bug in minimum transmission bitrate that made it ignore REMBs. A regression test has been added to catch it. BUG=3014 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5723 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19Fix a deadlock in ViEEncoder::DeliverFrame.wuchengli@chromium.org
The deadlock can happen when using HW encoder. HW encoder calls the encode complete callback on libjingle worker thread instead of ViECaptureThread. The capture thread can hold VieEncoder::|data_cs_| and wait for ModuleRtpRtcpImpl::|critical_section_module_ptrs_|. When libjingle worker thread runs encode complete callback, it can hold ModuleRtpRtcpImpl::|critical_section_module_ptrs_| and wait for VieEncoder::|data_cs_|. |default_rtp_rtcp_| is not guarded by |data_cs|. So move it out of the critical section to avoid the deadlock. BUG=chromium:352567 TEST=Run apprtc loopback on CrOS. Run apprtc between CrOS and Linux. Run vie_auto_test. R=henrik.lundin@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5721 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13Implement minimum transmit bitrate.pbos@webrtc.org
Utilizing minimum transmission bitrate prevents low remote bitrate estimates (bitrate estimation dips) when encoding non-complex content such as screenshare of a static image even though there's nothing wrong with the link. Requires pacing to be enabled for now, pending issue 3036. BUG=3014 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5694 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-07Avoid crash in ViEEncoder::DeRegisterExternalEncoder().fischman@webrtc.org
BUG=chromium:348222 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9519004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5660 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-06Adding a new ramp-up-down-up testhenrik.lundin@webrtc.org
The new test is based upon the exisiting rampup test, but also adds a low-rate period. The main purpose of the test is to verify the SuspendBelowMinBitrate functionality, which must be enabled for the test to pass. The CL also adds a change to the min bitrate in the send-side bandwidth estimator when SuspendBelowMinBitrate is enabled. An anonymous namespace is added around the StreamObserver classes in the test to avoid silent linker conflicts that could happen otherwise. Note: this CL depends on https://webrtc-codereview.appspot.com/9049004/ BUG=2636 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5646 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-27Adds APIs for reporting pacer queuing delay.jiayl@webrtc.org
BUG=2775 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8959005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-24Fix to get total number of sent and received rtcp packets.asapersson@webrtc.org
BUG=2638 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8979005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5601 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20Updated WebRTC version to 3.50elham@webrtc.org
TBR= wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8979004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5589 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20Modified overuse detection thresholds.asapersson@webrtc.org
BUG=1577 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8949005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5585 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19Add RTCP packet type counter (for getting statistics such as sent/received ↵asapersson@webrtc.org
NACK and FIR). Add counter to RTCP sender and RTCP receiver. Add video api GetRtcpPacketTypes(). BUG=2638 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18Remove external encryption API for VoE.solenberg@webrtc.org
BUG= R=henrika@webrtc.org, henrikg@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-17Reset estimate if no frame has been seen for a certain time (to avoid large ↵asapersson@webrtc.org
jitter if stop sending). Add delay before start processing after a reset. BUG=1577 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8699006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5561 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14Adding a critical section missing in r5543.stefan@webrtc.org
This fixes a race caught by the linux tsan bot. R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8739004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5551 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13Increase overuse and normal use thresholds for Mac.asapersson@webrtc.org
BUG=1577 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5544 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13Fixes a race when writing to send_padding_.stefan@webrtc.org
TEST=trybots R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8619004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5543 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12Set pacing bitrates in SetEncoder.pbos@webrtc.org
Before the change no padding was allowed before the first remote bitrate estimation was received. This bitrate estimation is based on what's actually sent. In tests I set codec.startBitrate to 300 instead of 325, which incidentally means that only the first layer gets encoded. As we only send ~150kbps instead of 300, the first REMB will significantly pull down the remote bitrate estimate instead of keeping the existing rate, even though there's no problem with the link. This was detected in RampUpTest.PacingWithRtx, (send_config.codec.startBitrate=300), which caused the tests to become a lot slower, and flake out more. By allowing padding initially we're able to keep our initial bitrate estimate. R=stefan@webrtc.org TEST=Running RampUpTest.WithPacingAndRtx with startBandwidth=300. BUG= Review URL: https://webrtc-codereview.appspot.com/8529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5534 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11Remove ViE external encryption API.solenberg@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8079005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5525 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10Add stats of incoming frame delays for debugging bandwidth estimation.jiayl@webrtc.org
BUG=crbug/338380 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29Connect webrtc::Config to WrappingBitrateEstimatorhenrik.lundin@webrtc.org
This is the second CL for this change. Connection to the ViE API remains to be done. BUG=2698 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5455 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..."mallinath@webrtc.org
We reverted the r5421 to allow us roll webrtc to chrome without any modifications to libjingle. Since webrtc is rolled with r5444, we can add back the original CL and changes to libjingle will be upstreamed in the next roll. TBR=andresp@webrtc.org > Revert 5421 "Fix deadlock on register/unregister observer while ..." > > Failure to compile on Chromium Internal bots, because of API changes. > > http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio > > You need to follow the steps mentioned in > https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer. > > Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs > as mentioned in the doc. > > > Fix deadlock on register/unregister observer while there is a an going callback. > > > > BUG=2835 > > R=mallinath@webrtc.org > > > > Review URL: https://webrtc-codereview.appspot.com/7119005 > > TBR=andresp@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/7679004 TBR=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7729005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5453 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-27Revert 5421 "Fix deadlock on register/unregister observer while ..."mallinath@webrtc.org
Failure to compile on Chromium Internal bots, because of API changes. http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio You need to follow the steps mentioned in https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer. Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs as mentioned in the doc. > Fix deadlock on register/unregister observer while there is a an going callback. > > BUG=2835 > R=mallinath@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/7119005 TBR=andresp@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23Fix deadlock on register/unregister observer while there is a an going callback.andresp@webrtc.org
BUG=2835 R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7119005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23Add callbacks for receive channel RTP statisticssprang@webrtc.org
This allows a listener to receive new statistics (byte/packet counts, etc) as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable. The change is primarily targeted at the new video engine API. TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up. BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-20Add configuration and test for extended RTCP reference time reports to new ↵asapersson@webrtc.org
video api. R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14Roll Chromium 238260 -> 243863wjia@webrtc.org
R=andrew@webrtc.org, henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5385 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13Updated Webrtc version to 3.49elham@webrtc.org
R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7049004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5374 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13Removes usage of ListWrapper from several files.henrike@webrtc.org
BUG=2164 R=andrew@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6269004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5373 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07Wire up statistics in video send stream of new video engine apisprang@webrtc.org
Note, this CL does not contain any tests. Those are implemeted as call tests and will be submitted when the receive stream is wired up as well. BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5559006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20Add thread_annotations for clang targets.andresp@webrtc.org
TESTED: As expected clang bots catched a few issues which are fixed with this CL, other bots ignore the annotations and compile fine. R=niklas.enbom@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6209004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5328 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20If the configured start bitrate is higher than the configures maxmflodman@webrtc.org
bitrate, cap the star rate accordingly. BUG=2720 R=asapersson@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6239004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5327 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20Race condition in ViECapturer::RegisterObserversprang@webrtc.org
Critical section ViECapturer.observer_cs_ should be taken when registering an observer. BUG=2734 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5326 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-19Update WebRTC to version 3.48tnakamura@webrtc.org
R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6189004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5324 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-19Add callbacks for receive channel RTCP statistics.sprang@webrtc.org
This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable. The change is primarily targeted at the new video engine API. TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up. BUG=2235 R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18Integrate fake_network_pipe into direct_transport.stefan@webrtc.org
TEST=trybots R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5321 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18Remove media_file from VideoEngine dependencies.pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6139004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16Revert r5294 to re-roll r5293.pbos@webrtc.org
To fix races in test each stream now owns its own encoder/decoder. R=mflodman@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/5919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-15Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."turaj@webrtc.org
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. > > BUG= > R=mflodman@webrtc.org, stefan@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/5409004 TBR=solenberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13Auto instantiate RBE depending on whether AST or TOF is available in ↵solenberg@webrtc.org
incoming packet stream. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5409004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13Callback for send bitrate estimates - new rollsprang@webrtc.org
Issue https://webrtc-codereview.appspot.com/4459004/ was commited as r5259, after which flakiness was detected and a rollback was performed at r5261. Patch Set 1 of this issue is the code submitted in r5259. Subsequent patch sets fixes a race condition which caused the seen problems. The root cause was a dead lock between a thread sending rtp packets and and a timed module processing thread: webrtc::RTPSender::BitrateUpdated() // Get RTPSender stats lock webrtc::Bitrate::Process() // Get Bitrate lock webrtc::RTPSender::ProcessBitrate() webrtc::ModuleRtpRtcpImpl::Process() ... webrtc::Bitrate::Update() // Get Bitrate lock webrtc::RTPSender::UpdateRtpStats() // Get RTPSender stats lock webrtc::RTPSender::SendToNetwork() ... This is fixed in Bitrate::Process() by releasing the lock before calling the callback. BUG=2235 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5619004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5281 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13Make sure channels in the same call are in the same channel group.mflodman@webrtc.org
Tested manually. I'll make a follow CL with a proper test once review.webrtc.org/5619004 has been committed. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5280 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13Making RemoteRateControl::min_configured_bit_rate_ configurablehenrik.lundin@webrtc.org
The minimum bitrate can now be configured from WrappingBitrateEstimator. BUG=2698 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5279 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13Update talk to 58127566 together withwu@webrtc.org
https://webrtc-codereview.appspot.com/5309005/. R=mallinath@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12Revert 5274 "Update talk to 58113193 together with https://webrt..."wu@webrtc.org
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. > > R=mallinath@webrtc.org, niklas.enbom@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/5719004 TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12Update talk to 58113193 together with ↵wu@webrtc.org
https://webrtc-codereview.appspot.com/5309005/. R=mallinath@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12Complete rewrite of demo application.henrike@webrtc.org
BUG=2122 R=andrew@webrtc.org, fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5273 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12Remove overloaded CpuOveruseMeasure function.asapersson@webrtc.org
R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5199005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5272 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11Add SwapFrame() to VideoSendStreamInput.pbos@webrtc.org
Optionally prevents doing a frame copy when putting frames into a VideoSendStream. PutFrame() is still there, which copies the frame. Also removes time_since_capture_ms as a parameter, since I420VideoFrame::render_time_ms() denotes when the frame was captured. BUG=2657 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-26 14:30:40 +0000