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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
2014-03-20Use codec width/height as the encoded_image width/height.wu@webrtc.org
The raw_->w and raw_->h which are the stored image width/height may not be the encoded image size in the case when the incoming frame has a odd size. R=marpan@google.com, marpan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10289004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5739 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20Changing the buffer size (slots) to 1.5 seconds @ 30 ms packetshenrik.lundin@webrtc.org
This is a relanding of r5725, now with a fix for the failing tests. BUG=2935 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10339005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5738 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
2014-03-20Prevent playout delay wrap-around in VoiceEnginehenrik.lundin@webrtc.org
In the case where a network glitch causes a packet to arrive so late that the jitter buffer has gone into expand mode, the playout timestamp could have been increased to a value that is larger than the RTP timestamp of the late packet when it finally arrives. This causes the difference to be negative, and would make the value wrap (unsigned). With this fix, the difference is set to zero when the playout timestamp is ahead of the incoming RTP timestamp. Further down in the method, a zero-value will lead to the averaging filter not being updated. BUG=3080 R=henrika@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5735 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20Removes error printout in voe_cmd_test which was caused by attempts to ↵henrika@webrtc.org
transmit RTCP packets even if a transport object was not registered. BUG=none R=xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5734 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20Extend perf tests to perform rampup on single stream.andresp@webrtc.org
R=kjellander@webrtc.org, stefan@webrtc.org BUG=3065 Review URL: https://webrtc-codereview.appspot.com/10049004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5733 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20Adjust the captured window rect when the window is maximized.jiayl@webrtc.org
GetWindowRect includes the window frames for maximized window even they are off screen, causing content outside the window being captured falsely. The fix is to remove the left/right/bottom window frame from the captured rect. Mouse capturing is adjusted accordingly as well. BUG=3076 R=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/10149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5732 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19Properly account for retransmitted packets when not using the pacer.stefan@webrtc.org
This regression was introduced in r5728. TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10269004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5729 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
2014-03-19Disabling SendsSetSimulcastSsrcs.pbos@webrtc.org
Disabling as bots are turning red. This should be because VideoSendStream::ReconfigureVideoCodec caps video_codec.startBitrate to max bitrates and as the start bitrate is just enough to transmit there might be some rounding errors here causing the top stream not to be sent. Since no REMB is received (send-side test) this remains as the transmit bitrate. I need some more time to figure out if this is the case so I'm disabling these for now to avoid reverting the big CL. VideoSendStreams aren't used in production yet. TBR=mflodman@webrtc.org BUG=3078 Review URL: https://webrtc-codereview.appspot.com/10229005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5727 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19Revert "Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets"henrik.lundin@webrtc.org
Build bots turned red. TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packetshenrik.lundin@webrtc.org
BUG=2935 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5725 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19Disable flaky CanSwitchToUseAllSsrcs.pbos@webrtc.org
Test flakes on bots, disabling while investigating. R=minyue@webrtc.org TBR=mflodman@webrtc.org BUG=3078 Review URL: https://webrtc-codereview.appspot.com/10119006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5724 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
2014-03-19Remove internal codecs from VideoSendStream.pbos@webrtc.org
Replaces VideoCodec in VideoSendStream::Config with an EncoderSettings struct. The EncoderSettings struct uses an external encoder for all codecs. This means that external users, such as libjingle, will provide the encoders themselves, removing the previous distinction of internal and external codecs. For now VideoSendStream translates to VideoCodec internally. In the interrim (before the corresponding change is implemented in VideoReceiveStream) tests convert EncoderSettings to VideoCodecs. Removes Call::GetVideoCodecs(). Disables RampUpTest.WithPacingAndRtx as its further exposed with changes to bitrates used in tests. BUG=2854,2992 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5722 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
2014-03-18Adds a method to WindowCapturer to bring a window to the front.jiayl@webrtc.org
BUG=3071 R=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/10079004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5718 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18Adding thread annotations to NetEq4henrik.lundin@webrtc.org
R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5716 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18Add #include <cstdlib> for std::abs.pbos@webrtc.org
IWYU violation. Fixes a breakage in the libc++ build of Chromium. BUG= R=earthdok@chromium.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5715 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18Resolves TSan v2 warnings in voe_auto_test.henrika@webrtc.org
See bug report for details. BUG=1590 R=tommi@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9859004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5714 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18Re-comitting r5711: "Fixing a flaky test in video_engine_tests"henrik.lundin@webrtc.org
The CL was reverted in r5712, due to bots going red. However, these bots are unrelated to this CL. Original description: VideoSendStreamTest.SuspendBelowMinBitrate was flaky. The problem was that when the first non-padding packet was sent after the stream was resumed, the statistics had not always been updated so that stats.suspended was false. After seeing the first non-padding packet after suspension, the test will now go into a state where it waits for the statistics to be changed. BUG=3068 R=pbos@webrtc.org TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10099004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5713 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18Revert 5711 "Fixing a flaky test in video_engine_tests"turaj@webrtc.org
> Fixing a flaky test in video_engine_tests > > VideoSendStreamTest.SuspendBelowMinBitrate was flaky. The problem was that when the first non-padding packet was sent after the stream was resumed, the statistics had not always been updated so that stats.suspended was false. After seeing the first non-padding packet after suspension, the test will now go into a state where it waits for the statistics to be changed. > > BUG=3068 > R=pbos@webrtc.org > TBR=stefan@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/10069004 TBR=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10089005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5712 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-17Fixing a flaky test in video_engine_testshenrik.lundin@webrtc.org
VideoSendStreamTest.SuspendBelowMinBitrate was flaky. The problem was that when the first non-padding packet was sent after the stream was resumed, the statistics had not always been updated so that stats.suspended was false. After seeing the first non-padding packet after suspension, the test will now go into a state where it waits for the statistics to be changed. BUG=3068 R=pbos@webrtc.org TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10069004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5711 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-17Small refactor on send_side_bandwidth_estimation.andresp@webrtc.org
R=stefan@webrtc.org BUG=3065 Review URL: https://webrtc-codereview.appspot.com/10029005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5710 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-17Refactor rampup tests:andresp@webrtc.org
- Cleanup test done condition (should be the same but with less code). - Split up functions blocks inside methods that were large. R=stefan@webrtc.org BUG=3065 Review URL: https://webrtc-codereview.appspot.com/10029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5708 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-17Tool to establish a loopback call via apprtc turn server.andresp@webrtc.org
For now the test keeps track of video bandwidth estimation and plots it using google visualization libraries after the test is concluded. There is also scripts to run a test and record the tcpdump. BUG=3037 R=hta@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5707 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14References to includes in third_party should be relative, not absolute.sprang@webrtc.org
BUG= R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5704 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14Add support for YUV4MPEG file reading to tools files. (Minor fix).mcasas@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5703 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14Add support for YUV4MPEG file reading to tools files.mcasas@webrtc.org
This CL adds support for reading .y4m files to the infra in video_quality_analysis.cc, adding new functions ExtractFrameFromYuvFile() and ExtractFrameFromY4mFile(), instad of the previous ExtractFrameFromI420(). The decision as to which one to use is taken from the file extension, if it is .y4m then is considered a YUV4MPEG file, otherwise is taken as a raw .yuv file. It also removes the pseudo duplicated function GetNextI420Frame(), that is used from psnr_ssim_analyzer.c, and adds support for y4m files there. Tested/validated via local compile-run. YUV4MPEG is a trivial container with a file header and a per-frame header, see [1] [1] http://wiki.multimedia.cx/index.php?title=YUV4MPEG2 BUG=https://code.google.com/p/chromium/issues/detail?id=343504 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5702 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14Fix a bug where network freeze during CNG causes delayhenrik.lundin@webrtc.org
Wrote a new NetEq unit test to test a network freeze during comfort noise playout. The network freezes and resumes during the silence period, and then resumes speech. It was verified that the delay increased due to the freeze, and this CL contains a fix for that problem. BUG=2995 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9849004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5701 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-14Remove legacy weirdness in Merge::Downsamplehenrik.lundin@webrtc.org
In practice, this will have only marginal effect. The length_limit was increased from 6.7 ms to 10 ms. This is compared with the input_length, which is equal to the decoded frame size. Thus, this change will only affect encoded frame sizes in this range (including 10 ms). BUG=2696 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5700 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13Stopping network threads before tearing down testhenrik.lundin@webrtc.org
Also initializing suspended_in_stats_ to false. TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9959005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5698 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13Race condition in RTPSendersprang@webrtc.org
RTPSender::sending_media_ should be guarded by send_critsect_. Fix this in GetSendSideDelay, SendPadData and TimeToSendPadding. Also add appropriate thread annotations. BUG=3029 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5697 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13Add max delay to trace based filters and enhances drop tail queues with ↵stefan@webrtc.org
delay statistics. R=solenberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9479004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5696 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13Re-landing "Routing SuspendChange to VideoSendStream::Stats"henrik.lundin@webrtc.org
This was originally committed as r5687, but reverted due to a flaky test. BUG=3040 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5695 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
2014-03-13Enable all RampUpTest.UpDownUp* testshenrik.lundin@webrtc.org
With issue 2987 fixed, all these tests can be enabled without problems. BUG=3010 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5693 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13Replace labs with std::abs.pbos@webrtc.org
Resolves clang 3.5 warnings on OS X for -Wabsolute-value. BUG=chromium:351479 R=andrew@webrtc.org, thakis@chromium.org Review URL: https://webrtc-codereview.appspot.com/9869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5692 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13Disable all protobuf dependent targets when enable_protobuf=0.andrew@webrtc.org
BUG=3045 TESTED=builds now when enable_protobuf=0 and modules_unittests still includes ApmTest.* when enable_protobuf=1. R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5690 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-12Enable VS2013 for Windows compilation by default.kjellander@webrtc.org
BUG=chromium:340973 TEST=All trybots passing runhooks and compile step (needs clobber flag -c to ensure old compile output doesn't cause it to fail). I also ran all the tests for the Windows trybots, which passed. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5687 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-12Remove platform-specific code from new-API tests.pbos@webrtc.org
We've had problems that seem to manifest in run_tests.mm getting stuck on exit. For our automated test targets only full_stack.cc was making use of the platform-specific renderers provided by webrtc_test_common and since no one currently monitors these the use case is hypothetical. Readding platform-specific renderers to video_loopback is tracked with issue 3039, though as far as I'm aware no one's currently using the video_loopback target. BUG=2987 R=kjellander@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9789004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5686 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-12Implement a test for an old corner-case in NetEqhenrik.lundin@webrtc.org
This CL implements a unit test to cover an case where comfort noise packets should be discarded. The situation arises when NetEq gets a duplicate comfort noise packet. Without this check, the duplicate would be decoded, and a the timing would shift. As it turned out, the corner-case funcionality was not completely accurate in NetEq4. This is because decision_logic_::cng_state_ is set after the corner-case check. In the old NetEq3, the corresponding state was changed before the check. This is now fixed. R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9639005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5685 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-12Developing NetEqImpl unit testshenrik.lundin@webrtc.org
Adding option to use mock or real objects instead of mocks. This will help future testing efforts, where each test case can select whether a mock or a real object should be used. Adding new test InsertPacketsUntilBufferIsFull. Removing a few uniteresting mock call warning. R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5684 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-11Disable TestOpusNewACM on Android.andrew@webrtc.org
It crashes flakily. TBR=tlegrand BUG=3006 Review URL: https://webrtc-codereview.appspot.com/9809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5682 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-11Revert "Routing SuspendChange to VideoSendStream::Stats"henrik.lundin@webrtc.org
The test VideoSendStreamTest.SuspendBelowMinBitrate seems flaky. Reverting and investigating. BUG=3040 Review URL: https://webrtc-codereview.appspot.com/9799004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5681 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-11Reorder includes in audio_processing_impl_unittest.andrew@webrtc.org
TBR=aluebs Review URL: https://webrtc-codereview.appspot.com/9779005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5680 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-11Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ ↵braveyao@webrtc.org
instead of _rtpRtcpModule now. BUG=3012 TEST=auto test R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5679 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-11Routing SuspendChange to VideoSendStream::Statshenrik.lundin@webrtc.org
Also checking that the statistics are properly updated in VideoSendStreamTest.SuspendBelowMinBitrate. Adding a test to SendStatisticsProxyTest. Checking callback status in rampup test, too. BUG=2457 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5678 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-10Classes and tests for audio an classifier. The class can be used to classify ↵jan.skoglund@webrtc.org
whether a frame of audio contains speech or music. The classifier uses the music/speech classifier in Opus. R=andrew@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5549004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5677 4adac7df-926f-26a2-2b94-8c16560cd09d
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-26 11:22:52 +0000