Commit Graph

96 Commits

Author SHA1 Message Date
c964d0b3fa Fixing some case-sensitive codec name comparisons.
As specified in RFC 4288, Section 4.2, and RFC 4855, Section 3, these
names should be case-insensitive. They already were being treated as
case-insensitive in some other places.

This bug was resulting in either a crash or no decoded video, depending
on the platform.

BUG=webrtc:6439, webrtc:7027

Review-Url: https://codereview.webrtc.org/2782273002
Cr-Commit-Position: refs/heads/master@{#17515}
2017-04-03 17:03:35 +00:00
36e9eb4773 Do not report quality limited resolution stats when quality scaler is disabled.
WebRTC.Video.QualityLimitedResolutionInPercent is reported as zero for calls when the quality scaler is off (and the degradation preference allows scaling).

Update SetResolutionRestrictionStats to specify if quality scaler is enabled.

BUG=webrtc:7432

Review-Url: https://codereview.webrtc.org/2783213002
Cr-Commit-Position: refs/heads/master@{#17487}
2017-03-31 12:29:12 +00:00
f7f7fb9cf8 Add traces for some video events.
This CL adds traces to compute the following metrics that getStats()
captures for video:
- googFrameRateSent
- googFrameRateInput
- googFirsReceived
- googNacksReceived
- googFrameWidthSent
- googFrameHeightSent
- googAvgEncodeMs
- googEncodeUsagePercent

BUG=chromium:653087

Review-Url: https://codereview.webrtc.org/2746903004
Cr-Commit-Position: refs/heads/master@{#17260}
2017-03-15 17:45:49 +00:00
93e1e23537 Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call.
Intervals when video is paused is no longer included in the stats:
"WebRTC.Video.BitrateSentInKbps"
"WebRTC.Video.MediaBitrateSentInKbps"
"WebRTC.Video.PaddingBitrateSentInKbps"
"WebRTC.Video.RetransmittedBitrateSentInKbps"
"WebRTC.Video.RtxBitrateSentInKbps"
"WebRTC.Video.FecBitrateSentInKbps"

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2536613002
Cr-Commit-Position: refs/heads/master@{#16447}
2017-02-06 13:18:35 +00:00
3d200bd6ac Remove FlexfecConfig and replace with specific struct in VideoSendStream.
The existence of FlexfecConfig is due to a naive design. Now when it
is not used on the receiving side (see https://codereview.webrtc.org/2542413002),
it is time to remove it from the sending side as well.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2621573002
Cr-Commit-Position: refs/heads/master@{#16097}
2017-01-16 14:59:19 +00:00
66d4b37414 Move histogram for number of pause events to per stream:
"WebRTC.Call.NumberOfPauseEvents" -> "WebRTC.Video.NumberOfPauseEvents"

Recorded if a certain time has passed (10 sec) since the first media packet was sent.

Moved to per stream to know when media has started and to prevent logging stats for calls that was never in use.

Add histogram for percentage of paused video time for sent video streams:
"WebRTC.Video.PausedTimeInPercent"

BUG=b/32659204

Review-Url: https://codereview.webrtc.org/2530393003
Cr-Commit-Position: refs/heads/master@{#15681}
2016-12-19 14:50:53 +00:00
0cd27ba088 Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ )
Reason for revert:
Bug affecting perf tests has been fixed. The issue was that I had accidentally disabled cpu overuse adaptation based on the encoders ScalingSettings, not just quality-based scaling.

Original issue's description:
> Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ )
>
> Reason for revert:
> Breaks perf tests
>
> Original issue's description:
> > Properly report number of quality downscales in stats.
> >
> > A regression was introduced in 876222f that caused these stats to
> > be reported incorrectly. This used to be only implemented for VP8
> > but should now be available for all codecs.
> >
> > BUG=webrtc:6860
> >
> > Review-Url: https://codereview.webrtc.org/2564373002
> > Cr-Commit-Position: refs/heads/master@{#15673}
> > Committed: 0c8c538835
>
> TBR=asapersson@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6860
>
> Review-Url: https://codereview.webrtc.org/2586783003
> Cr-Commit-Position: refs/heads/master@{#15678}
> Committed: fe04bd43cc

TBR=asapersson@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6860

Review-Url: https://codereview.webrtc.org/2588743002
Cr-Commit-Position: refs/heads/master@{#15680}
2016-12-19 14:32:16 +00:00
fe04bd43cc Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ )
Reason for revert:
Breaks perf tests

Original issue's description:
> Properly report number of quality downscales in stats.
>
> A regression was introduced in 876222f that caused these stats to
> be reported incorrectly. This used to be only implemented for VP8
> but should now be available for all codecs.
>
> BUG=webrtc:6860
>
> Review-Url: https://codereview.webrtc.org/2564373002
> Cr-Commit-Position: refs/heads/master@{#15673}
> Committed: 0c8c538835

TBR=asapersson@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6860

Review-Url: https://codereview.webrtc.org/2586783003
Cr-Commit-Position: refs/heads/master@{#15678}
2016-12-19 14:17:30 +00:00
0c8c538835 Properly report number of quality downscales in stats.
A regression was introduced in 876222f that caused these stats to
be reported incorrectly. This used to be only implemented for VP8
but should now be available for all codecs.

BUG=webrtc:6860

Review-Url: https://codereview.webrtc.org/2564373002
Cr-Commit-Position: refs/heads/master@{#15673}
2016-12-19 13:04:40 +00:00
876222f77d Move usage of QualityScaler to ViEEncoder.
This brings QualityScaler much more in line with OveruseFrameDetector.
The two classes are conceptually similar, and should be used in the
same way. The biggest changes in this CL are:
- Quality scaling is now only done in ViEEncoder and not in each
  encoder implementation separately.
- QualityScaler now checks the average QP asynchronously, instead of
  having to be polled on each frame.
- QualityScaler is no longer responsible for actually scaling the frames,
  but has a callback to ViEEncoder that it uses to express it's desire
  for lower resolution.

BUG=webrtc:6495

Review-Url: https://codereview.webrtc.org/2398963003
Cr-Commit-Position: refs/heads/master@{#15286}
2016-11-29 09:44:22 +00:00
320e45ad87 Use RateCounter for input/sent fps stats. Reports average of periodically computed stats over a call.
Intervals when video is paused is no longer included in the stats:
"WebRTC.Video.InputFramesPerSecond"
"WebRTC.Video.SentFramesPerSecond"

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2536743002
Cr-Commit-Position: refs/heads/master@{#15285}
2016-11-29 09:40:46 +00:00
a6a699a130 Sent bitrate stats are incorrect if FlexFEC is configured:
WebRTC.Video.BitrateSentInKbps
WebRTC.Video.MediaBitrateSentInKbps
WebRTC.Video.PaddingBitrateSentInKbps
WebRTC.Video.RetransmittedBitrateSentInKbps
WebRTC.Video.FecBitrateSentInKbps

RtpSender has two StreamDataCounters: for the non-RTX and the RTX stream.
The same counter (for the non-RTX stream) is reported for both the media SSRC and the FlexFEC SSRC.
Bitrate stats are summed for all SSRCs, thus the counter for the non-RTX stream is counted twice.
Do not store the counter for the FlexFEC SSRC.

Do not include info from FlexFEC substreams in VideoSendStream::Stats::ToString (periodically logged during a call).

BUG=webrtc:6774

Review-Url: https://codereview.webrtc.org/2525293002
Cr-Commit-Position: refs/heads/master@{#15238}
2016-11-25 11:52:55 +00:00
cd188f6031 Make SendStatisticsProxy let through FlexFEC packets.
This is initial work to get the stats working for FlexFEC.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2498393002
Cr-Commit-Position: refs/heads/master@{#15089}
2016-11-15 16:22:00 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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