6718e97e73
Add encode and decode time to histograms stats:
...
- "WebRTC.Video.EncodeTimeInMs"
- "WebRTC.Video.DecodeTimeInMs"
BUG=chromium:488243
Review URL: https://codereview.webrtc.org/1250203002
Cr-Commit-Position: refs/heads/master@{#9630}
2015-07-24 07:21:02 +00:00
d89920b74a
Add resolution and fps stats to histograms:
...
- "WebRTC.Video.InputWidthInPixels"
- "WebRTC.Video.InputHeightInPixels"
- "WebRTC.Video.SentWidthInPixels"
- "WebRTC.Video.SentHeightInPixels"
- "WebRTC.Video.ReceivedWidthInPixels"
- "WebRTC.Video.ReceivedHeightInPixels"
- "WebRTC.Video.RenderFramesPerSecond"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1228393008
Cr-Commit-Position: refs/heads/master@{#9611}
2015-07-22 13:52:03 +00:00
24b4eda6f4
Add sent framerates to histogram stats:
...
"WebRTC.Video.InputFramesPerSecond",
"WebRTC.Video.SentFramesPerSecond".
BUG=488243
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1169543005 .
Cr-Commit-Position: refs/heads/master@{#9446}
2015-06-16 08:17:09 +00:00
20f3f942a0
Clear bitrate stats for unused SSRCs.
...
Prevents bug where transmitted bitrate was reported as higher than what
was actually sent, since unused RTP modules weren't updated to say that
they sent zero.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49979004
Cr-Commit-Position: refs/heads/master@{#9192}
2015-05-15 09:33:27 +00:00
f2f828374c
Use rtc::CriticalSection in webrtc/video/.
...
Removes heap allocation from CriticalSection creation.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50839004
Cr-Commit-Position: refs/heads/master@{#9126}
2015-05-01 14:25:53 +00:00
2b4ce3a501
Convert webrtc/video/ abort/assert to CHECK/DCHECK.
...
Also replaces NULL with nullptr. This gives nicer error messages and
keeps style consistent.
BUG=1756
R=magjed@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42879004
Cr-Commit-Position: refs/heads/master@{#8831}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8831 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-23 13:13:15 +00:00
af612d5e07
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
...
Original cl description:
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/ .
Patchset 1 contains the original patch after rebase.
Patshet 2 fix webrtc_perf_tests reported in chromium:465306
Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/
BUG=1128
R=magjed@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47629004
Cr-Commit-Position: refs/heads/master@{#8776}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 09:51:44 +00:00
d7452a0168
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
...
This reverts commit r8633.
Reason for revert: Performance regressions in browser_tests_new_vie and webrtc_perf_tests.
BUG=1128,chromium:465287,chromium:465306
TBR=pbos,mflodman,perkj
Review URL: https://webrtc-codereview.appspot.com/46549004
Cr-Commit-Position: refs/heads/master@{#8670}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8670 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 15:13:13 +00:00
bcead305a2
Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
...
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
BUG=1128
R=magjed@webrtc.org , mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46429004
Cr-Commit-Position: refs/heads/master@{#8633}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8633 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-06 12:38:22 +00:00
891d48393e
Wire up target_media_bitrate in VideoSendStream.
...
Also wires up target_enc_bitrate in WebRtcVideoEngine2.
BUG=1667,1788
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42479004
Cr-Commit-Position: refs/heads/master@{#8515}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8515 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 13:16:17 +00:00
3e6e271ec3
Implement CpuOveruseMetrics as callbacks.
...
Adds avg_encode_ms and encode_usage_percent in WebRtcVideoEngine2 and
corresponding stats to VideoSendStream::Stats.
BUG=1667, 1788
R=asapersson@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42429004
Cr-Commit-Position: refs/heads/master@{#8513}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8513 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 12:20:24 +00:00
09c77b95bb
Add decoder-timing stats to VideoReceiveStream.
...
Also breaks out SsrcStats from VideoReceiveStream::Stats as they don't
have that much overlap.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667, 1788
Review URL: https://webrtc-codereview.appspot.com/40819004
Cr-Commit-Position: refs/heads/master@{#8501}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8501 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 10:42:45 +00:00
49096de442
DCHECK send DataCountersUpdated for valid SSRCs.
...
Also updates RTPSender to not update RTX stats when RTX is disabled.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42399004
Cr-Commit-Position: refs/heads/master@{#8489}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8489 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-24 22:38:22 +00:00
1d0fa5d352
Add RtcpPacketTypeCounter stats to new API.
...
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667,1788
Review URL: https://webrtc-codereview.appspot.com/37489004
Cr-Commit-Position: refs/heads/master@{#8429}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 12:47:45 +00:00
ce4e9a3562
Refactor some receive-side stats.
...
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 13:50:16 +00:00
273a414b0e
Report encoded frame size in VideoSendStream.
...
Implements reporting transmitted frame size in WebRtcVideoEngine2.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=4033
Review URL: https://webrtc-codereview.appspot.com/33399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 15:23:21 +00:00
0bae1fab4a
Wire up bandwidth stats to the new API and webrtcvideoengine2.
...
Adds stats to verify bandwidth and pacer stats.
BUG=1788
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 14:05:29 +00:00
58e2d262fc
Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().
...
Fixes issues where statistics only was reported for the first stream if
configured with simulcast, and in case of RTX the reported statistics was
depending on the order of the report blocks.
Also fixes issues with multiple report blocks in the SendStatisticsProxy and the
RtcpStatisticsCallback. SendStatisticsProxy is now aware of RTX ssrcs, and the
RTCPReceiver is calling the RtcpStatisticsCallback with the correct SSRCs, and
not only the primary stream SSRC.
R=mflodman@webrtc.org , sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-14 15:10:49 +00:00
168f23faa5
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
...
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 13:44:02 +00:00
4ef438e2de
Remove the send-side cname getter APIs from voice and video engine.
...
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 09:55:30 +00:00
de1429e9ad
Add thread annotations to Call API.
...
Also constified a lot of pointers and reordered members to make
protected members more grouped together.
R=kjellander@webrtc.org , stefan@webrtc.org
BUG=2770
Review URL: https://webrtc-codereview.appspot.com/15399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 13:00:21 +00:00
b10363f3b6
Re-landing "Routing SuspendChange to VideoSendStream::Stats"
...
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@5695 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 13:31:21 +00:00
be39470203
Revert "Routing SuspendChange to VideoSendStream::Stats"
...
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@5681 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-11 17:13:14 +00:00
1598b80f52
Routing SuspendChange to VideoSendStream::Stats
...
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@5678 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-11 14:57:35 +00:00
09315705b9
Wire up statistics in video receive stream of new API
...
This CL includes Call tests that test both send and receive sides.
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 12:06:29 +00:00
ccd42840bc
Wire up statistics in video send stream of new video engine api
...
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@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 09:54:34 +00:00
