327d8babc8
Add DecodedImageCallback::Decoded() function with custom decode time value.
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On Android, we would like to use MediaCodec output buffers to hold decoded frames until they can be rendered to a texture. There can only be one texture buffer used at the same time and therefore the calculated decode time in VCMTiming will be wrong since that calculation will also include the time where the decoder waited for the upper layers (that depend on network jitter and actual render time) to release the frame.
This new method will be used in
https://codereview.webrtc.org/1422963003/
BUG=webrtc:4993
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1414693006 .
Cr-Commit-Position: refs/heads/master@{#10576}
2015-11-10 13:00:45 +00:00
98f53510b2
system_wrappers: rename interface -> include
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BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
f244760827
Add histograms for receive statistics:
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- decoded frames per second ("WebRTC.Video.DecodedFramesPerSecond")
- percentage of delayed frames to rendered ("WebRTC.Video.DelayedFramesToRenderer")
- average delay (of delayed frames) to renderer ("WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs")
BUG=crbug/419657
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7847 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 14:13:26 +00:00
0422100818
Fix data race in VCMTiming::ResetDecodeTime.
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Also thread annotating class.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6653 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 15:25:37 +00:00
66773a032a
Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.
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BUG=3111
TEST=try bots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6074 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-07 17:09:44 +00:00
ed4cb56575
Remove timestamp_extrapolator's dependency to Clock and vcm defines.
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TEST=existing tests
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6058 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-06 04:50:49 +00:00
34c5da6b5e
Cleaned up logging in video_coding.
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Converted all calls to WEBRTC_TRACE to LOG(). Also removed a large number of less useful logs.
BUG=3153
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5887 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-11 14:08:35 +00:00
37bb4974e7
Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals.
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R=juberti@google.com , mikhal@webrtc.org , stefan@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 23:59:45 +00:00
30fb7b83d5
Add a log message to see video delay break down
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Shows video delay in terms of:
1. Min playout delay
2. Jitter delay
3. Max decode time
4. Render delay
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1674004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4242 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-18 21:37:09 +00:00
adc64a7216
VCM/Timing: Setting clear names to members & methods
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R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1524004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4140 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-30 16:20:18 +00:00
2eaf98b38b
Refactor VCM/Timing.
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No changes in functionality.
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/1514004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4081 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-21 17:58:43 +00:00
9f557c140e
Improve wraparound handling in the render time extrapolator.
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This was actually working as intended, but as r3970 changed when render timestamps were extrapolated to when a frame was taken out for decoding, the wraparound could have happened in the Update() step before it had happened in the ExtrapolateLocalTime() step. This causes render timestamps to be generated 13 hours into the future.
TEST=trybots
BUG=1787
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1497004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4055 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17 12:55:07 +00:00
759b041019
Relanding r3952: VCM: Updating receiver logic
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BUG=r1734
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1433004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3970 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 16:36:00 +00:00
4ce19b1664
Revert r3952 "VCM: Updating receiver logic"
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TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1410005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3963 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-06 13:16:51 +00:00
d3cd565ecf
VCM: Updating receiver logic
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R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1363005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3952 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-03 17:54:18 +00:00
d35964a1ce
Fixing AV sync.
...
Increased 2 const to allow for a bigger difference in AV sync.
BUG=1711
Re-wrote the ComputeDelays to be readable and remove the possibilities of returning values lower than base_target_delay_ms
R=mflodman@webrtc.org , mikhal@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1367004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3922 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-30 16:06:10 +00:00
6faba6edc9
VCM: Setting buffering delay in timing
...
Review URL: https://webrtc-codereview.appspot.com/1338004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3921 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-30 15:39:34 +00:00
dbd6a6d653
Updating delay for first value
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/1327005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3865 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-17 16:23:22 +00:00
7b859cc1e9
Webrtc_Word32 => int32_t in video_coding/main/
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BUG=
Review URL: https://webrtc-codereview.appspot.com/1279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 15:54:38 +00:00
ef9f76a59d
Adding a receive side API for buffering mode.
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At the same time, renaming the send side API.
Review URL: https://webrtc-codereview.appspot.com/1104004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 23:22:18 +00:00
a678a3baee
Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
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TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1044004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-21 07:42:11 +00:00
14b43beb7c
Move src/ -> webrtc/
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TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00
