83 Commits

Author SHA1 Message Date
asapersson
827cab3fc2 Add qp counter for H264 in SendStatisticsProxy.
BUG=webrtc:6578

Review-Url: https://codereview.webrtc.org/2437323002
Cr-Commit-Position: refs/heads/master@{#14895}
2016-11-02 16:08:53 +00:00
perkj
803d97f159 Let ViEEncoder express resolution requests as Sinkwants.
This removes the VideoSendStream::LoadObserver interface and the implementation in WebrtcVideoSendStream and replace it with VideoSinkWants through the VideoSourceInterface.

To do that that, some stats for CPU adaptation is moved into VideoSendStream. Also handling of the CVO rtp header extension is moved to VideoSendStreamImpl.

BUG=webrtc:5687
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/2304363002
Cr-Commit-Position: refs/heads/master@{#14877}
2016-11-01 18:45:54 +00:00
sakal
87da404883 Implement qpSum stat for video send ssrc stats.
Implemented as defined by this pull request: https://github.com/w3c/webrtc-stats/pull/70

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2430603003
Cr-Commit-Position: refs/heads/master@{#14851}
2016-10-31 13:53:51 +00:00
sakal
43536c3d6a Implement framesEncoded stat in video send ssrc stats.
Implemented as defined by this pull request: https://github.com/w3c/webrtc-stats/pull/70

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2421193003
Cr-Commit-Position: refs/heads/master@{#14734}
2016-10-24 09:09:39 +00:00
brandtr
b5f2c3fbe9 Rename FecConfig to UlpfecConfig in config.h.
Also rename some related minor methods. No functional changes
are intended/expected.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2391963002
Cr-Commit-Position: refs/heads/master@{#14513}
2016-10-05 06:28:43 +00:00
Per
a48ddb7636 Add VideoSendStream::Stats::prefered_media_bitrate_bps
This cl move calculation of stats for prefered_media_bitrate_bps from webrtcvideoengine2.GetStats to SendStatisticsProxy::OnEncoderReconfigured.
This aligns better with how other send stats are reported and is needed as a prerequisite for moving video encoder configuration due to video resolution change
from WebRtcVideoEngine2 to ViEEncoder.

BUG=webrtc:6371
R=mflodman@webrtc.org, sprang@webrtc.org

Review URL: https://codereview.webrtc.org/2368223002 .

Cr-Commit-Position: refs/heads/master@{#14431}
2016-09-29 09:49:01 +00:00
asapersson
1d02d3e5e6 Remove RTC_LOGGED_* macro.
BUG=

Review-Url: https://codereview.webrtc.org/2326843003
Cr-Commit-Position: refs/heads/master@{#14174}
2016-09-10 05:40:34 +00:00
perkj
26091b1118 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
cl was originally reviewed here:
https://codereview.webrtc.org/2060403002/

- Add task queue to Call with the intent of replacing the use of one of the process threads.

- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.

- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.

- VideoEncoderConfig and VideoSendStream::Config support move semantics.

- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.

TBR=mflodman@webrtc.org
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2250123002
Cr-Commit-Position: refs/heads/master@{#14014}
2016-09-01 08:17:43 +00:00
perkj
275afc5d2e Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2253563002
Cr-Commit-Position: refs/heads/master@{#14012}
2016-09-01 07:21:18 +00:00
perkj
8eb37a39e7 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
Reason for revert:
Failed on Win 10 Chrome FYI.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3847/steps/content_browsertests/logs/stdio

#
# Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\base\task_queue_win.cc, line 138
# last system error: 87
# Check failed: ((DWORD)0xFFFFFFFF) != result (4294967295 vs. 4294967295)
#

WebRtcBrowserTest

#

Original issue's description:
> - Add task queue to Call with the intent of replacing the use of one of the process threads.
>
> - Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
>
> - BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
>
> - VideoEncoderConfig and VideoSendStream::Config support move semantics.
>
> - The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
>
> BUG=webrtc:5687
>
> Committed: https://crrev.com/cc168360f41322332860cb075edeb1cde21aa473
> Cr-Commit-Position: refs/heads/master@{#13767}

TBR=tommi@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,sprang@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2248713003
Cr-Commit-Position: refs/heads/master@{#13774}
2016-08-16 09:40:59 +00:00
perkj
cc168360f4 - Add task queue to Call with the intent of replacing the use of one of the process threads.
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.

- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.

- VideoEncoderConfig and VideoSendStream::Config support move semantics.

- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.

BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/2060403002
Cr-Commit-Position: refs/heads/master@{#13767}
2016-08-16 07:38:51 +00:00
asapersson
2e5cfcd6c2 Add periodic logging of video stats.
Add ToString method to: Call::Stats, VideoSendStream::Stats, VideoReceiveStream::Stats and log stats periodically (every 10 seconds).

BUG=

Review-Url: https://codereview.webrtc.org/2133073002
Cr-Commit-Position: refs/heads/master@{#13727}
2016-08-11 15:41:26 +00:00
asapersson
4374a09f9b Only update codec type histogram if lifetime is long enough (10 sec).
Add metrics for Call/VideoSendStream/VideoReceiveStream lifetime.

BUG=

Review-Url: https://codereview.webrtc.org/2136533002
Cr-Commit-Position: refs/heads/master@{#13537}
2016-07-27 07:39:17 +00:00
sprang
cd349d9743 Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
Reason for revert:
Upstream fixes in place, should be OK now.

Original issue's description:
> Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ )
>
> Reason for revert:
> Breaks upstream code.
>
> Original issue's description:
> > Refactor NACK bitrate allocation
> >
> > Nack bitrate allocation should not be done on a per-rtp-module basis,
> > but rather shared bitrate pool per call. This CL moves allocation to the
> > pacer and cleans up a bunch if bitrate stats handling.
> >
> > BUG=
> > R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
> >
> > Committed: 5fc59e810b
>
> TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/e5dd44101eca485f5ad12e5f7ce6f6b0d204116b
> Cr-Commit-Position: refs/heads/master@{#13417}

TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=

Review-Url: https://codereview.webrtc.org/2146013002
Cr-Commit-Position: refs/heads/master@{#13465}
2016-07-13 16:11:38 +00:00
aluebs
a49f1105eb Revert of Reland Issue 2061423003: Refactor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131313002/ )
Reason for revert:
It keeps breaking upstream.

Original issue's description:
> Reland Issue 2061423003: Refactor NACK bitrate allocation
>
> This is a reland of https://codereview.webrtc.org/2061423003/
> Which was reverted in https://codereview.webrtc.org/2131913003/
>
> The reason for the revert was that some upstream code used
> RtpSender::SetTargetBitrate(). I've added that back as a no-op until we
> it's been brought up to date.
>
> TBR=tommi@webrtc.org
>
> Committed: 05ce4ae31f

TBR=tommi@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2130423002
Cr-Commit-Position: refs/heads/master@{#13419}
2016-07-08 18:02:02 +00:00
Erik Språng
05ce4ae31f Reland Issue 2061423003: Refactor NACK bitrate allocation
This is a reland of https://codereview.webrtc.org/2061423003/
Which was reverted in https://codereview.webrtc.org/2131913003/

The reason for the revert was that some upstream code used
RtpSender::SetTargetBitrate(). I've added that back as a no-op until we
it's been brought up to date.

TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2131313002 .

Cr-Commit-Position: refs/heads/master@{#13418}
2016-07-08 17:11:23 +00:00
sprang
e5dd44101e Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ )
Reason for revert:
Breaks upstream code.

Original issue's description:
> Refactor NACK bitrate allocation
>
> Nack bitrate allocation should not be done on a per-rtp-module basis,
> but rather shared bitrate pool per call. This CL moves allocation to the
> pacer and cleans up a bunch if bitrate stats handling.
>
> BUG=
> R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
>
> Committed: 5fc59e810b

TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/2131913003
Cr-Commit-Position: refs/heads/master@{#13417}
2016-07-08 16:39:02 +00:00
Erik Språng
5fc59e810b Refactor NACK bitrate allocation
Nack bitrate allocation should not be done on a per-rtp-module basis,
but rather shared bitrate pool per call. This CL moves allocation to the
pacer and cleans up a bunch if bitrate stats handling.

BUG=
R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2061423003 .

Cr-Commit-Position: refs/heads/master@{#13416}
2016-07-08 16:15:29 +00:00
perkj
f5b2e519b4 Fix stats for encoder target bitrate when target rate is zero.
When the target bitrate is zero, currently  VideoSendStream.Stats.target_media_bitrate_bps show the last set rate before the target was set to zero.

BUG=webrtc::5687 b/29574845

Review-Url: https://codereview.webrtc.org/2122743003
Cr-Commit-Position: refs/heads/master@{#13386}
2016-07-05 15:34:08 +00:00
Per
69b332df83 Move logic for calculating needed bitrate overhead used by NACK and FEC to VideoSender.
This cl split the class MediaOptimization into two parts. One that deals with frame dropping and stats and one new class called ProtectionBitrateCalculator that deals with  calculating the needed FEC parameters and how much of the estimated network bitrate that can be used by an encoder

Note that the logic of how FEC and the needed bitrates is not changed.

BUG=webrtc:5687
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1972083002 .

Cr-Commit-Position: refs/heads/master@{#13018}
2016-06-02 13:45:53 +00:00
Honghai Zhang
82d7862fe7 Change default timestamp to 64 bits in all webrtc directories.
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1835053002 .

Cr-Commit-Position: refs/heads/master@{#12646}
2016-05-06 18:29:27 +00:00
perkj
376b192ea3 Remove VideoCodingModule::VCMPacketizationCallback
And move encoder name cb to VCMSendStatisticsCallback.

BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/1900193004
Cr-Commit-Position: refs/heads/master@{#12596}
2016-05-02 18:35:33 +00:00
kjellander
02b3d275a0 Reland of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #1 id:1 of https://codereview.webrtc.org/1903193002/ )
Reason for revert:
A fix is being prepared downstream so this can now go in.

Original issue's description:
> Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ )
>
> Reason for revert:
> API changes broke downstream.
>
> Original issue's description:
> > Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
> > EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
> > EncodedImageCallback can of course be cleaned up in the future.
> >
> > This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
> >
> > BUG=webrtc::5687
> >
> > Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb
> > Cr-Commit-Position: refs/heads/master@{#12436}
>
> TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5687
>
> Committed: https://crrev.com/a261e6136655af33f283eda8e60a6dd93dd746a4
> Cr-Commit-Position: refs/heads/master@{#12441}

TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687

Review URL: https://codereview.webrtc.org/1905583002

Cr-Commit-Position: refs/heads/master@{#12442}
2016-04-20 12:06:01 +00:00
kjellander
a261e61366 Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ )
Reason for revert:
API changes broke downstream.

Original issue's description:
> Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
> EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
> EncodedImageCallback can of course be cleaned up in the future.
>
> This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
>
> BUG=webrtc::5687
>
> Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb
> Cr-Commit-Position: refs/heads/master@{#12436}

TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc::5687

Review URL: https://codereview.webrtc.org/1903193002

Cr-Commit-Position: refs/heads/master@{#12441}
2016-04-20 11:13:30 +00:00
perkj
f5d55aaecd Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
EncodedImageCallback can of course be cleaned up in the future.

This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.

BUG=webrtc::5687

Review URL: https://codereview.webrtc.org/1897233002

Cr-Commit-Position: refs/heads/master@{#12436}
2016-04-20 08:17:11 +00:00
asapersson
5265fedffe Add histogram stats for average QP per frame for VP9 (for sent video streams):
- "WebRTC.Video.Encoded.Qp.Vp9"
- "WebRTC.Video.Encoded.Qp.Vp9.S0"
- "WebRTC.Video.Encoded.Qp.Vp9.S1"
- "WebRTC.Video.Encoded.Qp.Vp9.S2"

BUG=

Review URL: https://codereview.webrtc.org/1870043002

Cr-Commit-Position: refs/heads/master@{#12402}
2016-04-18 09:58:52 +00:00
Peter Boström
74f6e9ea23 Replace NULL with nullptr in webrtc/video.
BUG=
R=danilchap@webrtc.org

Review URL: https://codereview.webrtc.org/1855433002 .

Cr-Commit-Position: refs/heads/master@{#12218}
2016-04-04 15:56:22 +00:00
asapersson
118ef00594 Add histogram stats for average QP per frame for VP8 (for sent video streams):
- "WebRTC.Video.Encoded.Qp.Vp8"
- "WebRTC.Video.Encoded.Qp.Vp8.S0"
- "WebRTC.Video.Encoded.Qp.Vp8.S1"
- "WebRTC.Video.Encoded.Qp.Vp8.S2"

BUG=

Review URL: https://codereview.webrtc.org/1523293002

Cr-Commit-Position: refs/heads/master@{#12174}
2016-03-31 07:00:25 +00:00
asapersson
58d992e025 Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).
Adds logging of:
- video stats that are recorded when a stream is removed
- bitrate stats that are recorded at the end of a call
- initial bwe rampup stats

BUG=

Review URL: https://codereview.webrtc.org/1788783002

Cr-Commit-Position: refs/heads/master@{#12133}
2016-03-29 09:15:11 +00:00
Erik Språng
22c2b4814a Move RTP stats histograms from VieChannel to SendStatisticsProxy.
Also slice for screensharing.

BUG=
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1734933002 .

Cr-Commit-Position: refs/heads/master@{#11822}
2016-03-01 08:40:54 +00:00
sprang
07fb9be37f Move RTCP histograms from vie_channel to video channel stats proxies.
Also slice those histograms on content type.

BUG=

Review URL: https://codereview.webrtc.org/1720883002

Cr-Commit-Position: refs/heads/master@{#11748}
2016-02-24 15:55:06 +00:00
sprang
e2d83d6560 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()
Also move some stats reporting from vie_channel to send stats proxy

BUG=

Review URL: https://codereview.webrtc.org/1669623004

Cr-Commit-Position: refs/heads/master@{#11688}
2016-02-19 17:03:34 +00:00
Peter Boström
d1d66bab3d Remove ViEChannel calls for VideoReceiveStream.
Remove hops into ViEChannel for calls directly into RtpRtcp and
ViEReceiver from VideoReceiveStream.

Some calls are more complex and will be removed later.

BUG=webrtc:5494
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1671893002 .

Cr-Commit-Position: refs/heads/master@{#11526}
2016-02-08 13:07:22 +00:00
Peter Boström
e449915455 Measure encoding time on encode callbacks.
Permits measuring encoding time even when performed on another thread,
typically for hardware encoding, instead of assuming that encoding is
blocking the calling thread.

Permitted encoding time is increased for hardware encoders since they
can be timed to keep 30fps, for instance, without indicating overload.

Merges EncodingTimeObserver into EncodedFrameObserver to have one post-encode
callback.

BUG=webrtc:5042, webrtc:5132
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1569853002 .

Cr-Commit-Position: refs/heads/master@{#11499}
2016-02-05 10:13:41 +00:00
asapersson
c2148a50d2 Integrate helper macros for calling histograms with different names (real-time vs screenshare and rampup metrics).
Sparse macro is replaced and new implementation in metrics.h is used.

BUG=webrtc:5283

Review URL: https://codereview.webrtc.org/1564923008

Cr-Commit-Position: refs/heads/master@{#11483}
2016-02-04 08:33:29 +00:00
asapersson
28ba92731d Switch to use new implementation in metrics.h.
Sparse macro replaced for all video histograms that have a constant name.

BUG=webrtc:5283

Review URL: https://codereview.webrtc.org/1616153005

Cr-Commit-Position: refs/heads/master@{#11368}
2016-01-25 13:58:27 +00:00
Tommi
97888bd95a Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video.
While doing this, I made a couple of minor changes:
* Removed unused variables (one lock and one video frame variable)
* Switched over to a scoped lock in remb.cc and removed an if() in a function where we can just return the expression being checked.

BUG=
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1613053003 .

Cr-Commit-Position: refs/heads/master@{#11349}
2016-01-21 22:25:12 +00:00
asapersson
59bac1a4c5 Fix for stats updated twice when switching content type (realtime <-> screenshare). Add unittest.
BUG=

Review URL: https://codereview.webrtc.org/1543933004

Cr-Commit-Position: refs/heads/master@{#11180}
2016-01-08 07:36:06 +00:00
asapersson
53805324c0 Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates.
This implementation will be replaced by a faster one and sparse will be removed.

BUG=webrtc:5283

Review URL: https://codereview.webrtc.org/1530913002

Cr-Commit-Position: refs/heads/master@{#11099}
2015-12-21 09:46:25 +00:00
Peter Boström
b7d9a97ce4 Expose codec implementation names in stats.
Used to distinguish between software/hardware encoders/decoders and
other implementation differences. Useful for tracking quality
regressions related to specific implementations.

BUG=webrtc:4897
R=hta@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1406903002 .

Cr-Commit-Position: refs/heads/master@{#11084}
2015-12-18 15:01:23 +00:00
asapersson
17821db197 Wire up bandwidth limitation info to GetStats and adapt_reason.
The input resolution (output from video_adapter) can be further scaled down or higher video layer(s) can be dropped due to bitrate constraints.

BUG=webrtc:4112

Review URL: https://codereview.webrtc.org/1502173002

Cr-Commit-Position: refs/heads/master@{#11006}
2015-12-14 10:08:19 +00:00
asapersson
1aa420b6aa Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead.
BUG=

Review URL: https://codereview.webrtc.org/1278383002

Cr-Commit-Position: refs/heads/master@{#10911}
2015-12-07 11:12:27 +00:00
sprang
b4a1ae5299 Add separate send-side UMA stats for screenshare and video.
This CL duplicates all the histograms in SendStatisticsProxy. Might be
overkill, but we don't know which stats will be interesting and it makes
the change easier.

BUG=

Review URL: https://codereview.webrtc.org/1433393002

Cr-Commit-Position: refs/heads/master@{#10885}
2015-12-03 16:10:13 +00:00
asapersson
6f14be8df8 Add limit for minimum number of required samples before recording input and sent framerate stats.
BUG=

Review URL: https://codereview.webrtc.org/1446443002

Cr-Commit-Position: refs/heads/master@{#10644}
2015-11-16 08:40:57 +00:00
Tim Psiaki
ad13d2f817 Round Rate computations from RateTracker.
BUG=534221
R=asapersson@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1410533004 .

Cr-Commit-Position: refs/heads/master@{#10592}
2015-11-11 00:34:58 +00:00
asapersson
f040b2367d Add histograms for send-side delay stats for a sent video stream:
- "WebRTC.Video.SendSideDelayInMs"
- "WebRTC.Video.SendSideDelayMaxInMs"

BUG=chromium:512752

Review URL: https://codereview.webrtc.org/1405023014

Cr-Commit-Position: refs/heads/master@{#10502}
2015-11-04 08:59:10 +00:00
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include
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
asapersson
2a0a2a410f Add stats for used video codec type for a sent video stream:
- "WebRTC.Video.Encoder.CodecType"

BUG=chromium:512752

Review URL: https://codereview.webrtc.org/1426673002

Cr-Commit-Position: refs/heads/master@{#10423}
2015-10-27 08:32:06 +00:00
Peter Boström
415d2cd745 Use webrtc/base/logging.h for video.
BUG=webrtc:5118
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1415413004 .

Cr-Commit-Position: refs/heads/master@{#10403}
2015-10-26 10:35:26 +00:00
Peter Boström
49e196af40 Remove VideoFrameType aliases for FrameType.
No longer used in Chromium, so these can now be removed.

BUG=webrtc:5042
R=mflodman@webrtc.org
TBR=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1415693002 .

Cr-Commit-Position: refs/heads/master@{#10390}
2015-10-23 13:58:27 +00:00