74 Commits

Author SHA1 Message Date
brandtr
e950cadba5 Wire up FlexfecSender in RTP module and VideoSendStream.
FlexfecSender is owned and configured by VideoSendStream.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2501503003
Cr-Commit-Position: refs/heads/master@{#15082}
2016-11-15 13:25:44 +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
palmkvist
e75f204b06 Expose Ivf logging through the native API
BUG=webrtc:6300

Review-Url: https://codereview.webrtc.org/2303273002
Cr-Commit-Position: refs/heads/master@{#14419}
2016-09-28 13:19:53 +00:00
perkj
a49cbd3e24 Replace VideoCapturerInput with VideoSinkInterface.
Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)

This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values

This cl
Revert "Revert of Replace interface VideoCapturerInput with VideoSinkInterface. (patchset #13 id:280001 of https://codereview.webrtc.org/2257413002/ )"

This reverts commit 9fdbda6aa3f66ea872344c22e79b23361047cbab.

and fix the problem in the original cl in video_quality_test.cc

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

Review-Url: https://codereview.webrtc.org/2348533002
Cr-Commit-Position: refs/heads/master@{#14265}
2016-09-16 14:53:48 +00:00
perkj
9fdbda6aa3 Revert of Replace interface VideoCapturerInput with VideoSinkInterface. (patchset #13 id:280001 of https://codereview.webrtc.org/2257413002/ )
Reason for revert:
Fails on Mac and Linux webrtc_perf_tests

Original issue's description:
> Replace VideoCapturerInput with VideoSinkInterface.
> Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)
>
> This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values.
>
> BUG=webrtc:5687
> // Android CQ seems broken.
> NOTRY=true
>
> Committed: https://crrev.com/95a226f55ae7e32b83a6ba96232fb105a014dc6c
> Cr-Commit-Position: refs/heads/master@{#14238}

TBR=nisse@webrtc.org,sprang@webrtc.org,mflodman@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:5687

Review-Url: https://codereview.webrtc.org/2344923002
Cr-Commit-Position: refs/heads/master@{#14239}
2016-09-15 16:19:28 +00:00
perkj
95a226f55a Replace VideoCapturerInput with VideoSinkInterface.
Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)

This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values.

BUG=webrtc:5687
// Android CQ seems broken.
NOTRY=true

Review-Url: https://codereview.webrtc.org/2257413002
Cr-Commit-Position: refs/heads/master@{#14238}
2016-09-15 15:57:26 +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
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
pbos
1ba8d39a9c Remove webrtc/stream.h and unutilized inheritance.
Removes inheritance and a virtual call. Also removes a root header that
would have needed to be moved into a subdirectory otherwise to prevent
circular dependencies.

BUG=webrtc:4243
R=kjellander@webrtc.org, solenberg@webrtc.org
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/1924793002
Cr-Commit-Position: refs/heads/master@{#12586}
2016-05-02 03:18:36 +00:00
pbos
a96b60b3a6 Move frame_callback.h to common_video/include.
BUG=webrtc:4243
R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1902543002

Cr-Commit-Position: refs/heads/master@{#12419}
2016-04-19 04:12:57 +00:00
nisse
d30a1117f8 Change pre_encode_callback to get a const frame.
Used only by tests. Deleted the EndToEndTest.UsesFrameCallbacks, which
modified pixel data. Change callback from in EndToEndTest.GetStats to call SleepMs, rath than
modifying the timestamp.

BUG=

Review URL: https://codereview.webrtc.org/1891733002

Cr-Commit-Position: refs/heads/master@{#12406}
2016-04-18 12:15:33 +00:00
nisse
7ade7b3282 Delete class webrtc::VideoRenderer and its header file.
To replace the SmoothsRenderedFrames method, added a corresponding
flag to VideoReceiveStream::Config instead.

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1818023002

Cr-Commit-Position: refs/heads/master@{#12102}
2016-03-23 11:48:17 +00:00
Peter Boström
905f8e7e9d Make ReconfigureVideoEncoder void.
Also moves and simplifies SetSendCodec from VideoSendStream to mostly
inside ViEEncoder. This is necessary for making
ReconfigureVideoEncoder asynchronous as we don't post any result back.

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

Review URL: https://codereview.webrtc.org/1754283002 .

Cr-Commit-Position: refs/heads/master@{#11847}
2016-03-02 16:00:07 +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
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
deadbeef
1387149ad1 Adding reduced size RTCP configuration down to the video stream level.
Still waiting to turn on negotiation (in mediasession.cc)
until we verify it's working as expected.

BUG=webrtc:4868

Review URL: https://codereview.webrtc.org/1418123003

Cr-Commit-Position: refs/heads/master@{#10958}
2015-12-09 20:37:59 +00:00
ivica
8d15bd6dab Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
Reason for revert:
The reverted commit didn't affect the tests, but the one before: https://codereview.webrtc.org/1385563005/

I've run the test that was failing (EndToEndTest.AssignsTransportSequenceNumbers) locally multiple times, and it works fine (finishes successfully in 150-170ms).

Original issue's description:
> Revert of Collecting encode_time_ms for each frame (patchset #13 id:220001 of https://codereview.webrtc.org/1374233002/ )
>
> Reason for revert:
> Breaks EndToEndTest.AssignsTransportSequenceNumbers in video_engine_tests
> on several bots:
> http://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/5507
> http://build.chromium.org/p/client.webrtc/builders/Mac64%20Debug/builds/4815
> http://build.chromium.org/p/client.webrtc/builders/Win%20SyzyASan/builds/3272
> http://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4414
>
> It seems very unfortunate that it breaks on _exactly_ the bot configs that aren't covered by the CQ trybots.
>
> Original issue's description:
> > Collecting encode_time_ms for each frame.
> >
> > Also, in Sample struct, replacing double with the original type.
> > It makes more sense to save the original data as truthful as possible, and then
> > convert it to double later if necessary (in the plot script).
> >
> > Committed: https://crrev.com/092b13384e57b33e2003d9736dfa1f491e76f938
> > Cr-Commit-Position: refs/heads/master@{#10184}
>
> TBR=sprang@webrtc.org,pbos@webrtc.org,mflodman@webrtc.org,asapersson@webrtc.org,ivica@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/810447972425e890bc7911af27f894b86e9b7e6f
> Cr-Commit-Position: refs/heads/master@{#10185}

TBR=sprang@webrtc.org,pbos@webrtc.org,mflodman@webrtc.org,asapersson@webrtc.org,kjellander@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1390163002

Cr-Commit-Position: refs/heads/master@{#10195}
2015-10-07 09:43:25 +00:00
kjellander
8104479724 Revert of Collecting encode_time_ms for each frame (patchset #13 id:220001 of https://codereview.webrtc.org/1374233002/ )
Reason for revert:
Breaks EndToEndTest.AssignsTransportSequenceNumbers in video_engine_tests
on several bots:
http://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/5507
http://build.chromium.org/p/client.webrtc/builders/Mac64%20Debug/builds/4815
http://build.chromium.org/p/client.webrtc/builders/Win%20SyzyASan/builds/3272
http://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4414

It seems very unfortunate that it breaks on _exactly_ the bot configs that aren't covered by the CQ trybots.

Original issue's description:
> Collecting encode_time_ms for each frame.
>
> Also, in Sample struct, replacing double with the original type.
> It makes more sense to save the original data as truthful as possible, and then
> convert it to double later if necessary (in the plot script).
>
> Committed: https://crrev.com/092b13384e57b33e2003d9736dfa1f491e76f938
> Cr-Commit-Position: refs/heads/master@{#10184}

TBR=sprang@webrtc.org,pbos@webrtc.org,mflodman@webrtc.org,asapersson@webrtc.org,ivica@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1383283005

Cr-Commit-Position: refs/heads/master@{#10185}
2015-10-06 18:34:14 +00:00
ivica
092b13384e Collecting encode_time_ms for each frame.
Also, in Sample struct, replacing double with the original type.
It makes more sense to save the original data as truthful as possible, and then
convert it to double later if necessary (in the plot script).

Review URL: https://codereview.webrtc.org/1374233002

Cr-Commit-Position: refs/heads/master@{#10184}
2015-10-06 14:13:50 +00:00
pbos
2d566686a2 Unify Transport and newapi::Transport interfaces.
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1369263002

Cr-Commit-Position: refs/heads/master@{#10096}
2015-09-28 16:59:36 +00:00
solenberg
e526974759 Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1325263002

Cr-Commit-Position: refs/heads/master@{#9891}
2015-09-08 12:13:25 +00:00
sophiechang
47d78cc8ad Pass the encoder's internal source property through to video_sender to request a keyframe from the external encoder
BUG=

Review URL: https://codereview.webrtc.org/1263663005

Cr-Commit-Position: refs/heads/master@{#9853}
2015-09-04 01:24:53 +00:00
solenberg
4fbae2b791 Add send transports to individual webrtc::Call streams.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1273363005

Cr-Commit-Position: refs/heads/master@{#9807}
2015-08-28 11:07:15 +00:00
Jelena Marusic
cd6702282a Define Stream base classes
BUG=webrtc:4690

Defined classes Stream, SendStream and ReceiveStream. Inherited existing stream classes from either SendStream or ReceiveStream.
This is a step towards having a Transport associated with streams instead of a Call. It will allow a lot of code in the Call to be media type agnostic.

R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1226123005 .

Cr-Commit-Position: refs/heads/master@{#9591}
2015-07-16 07:30:20 +00:00
Peter Boström
4b91bd0897 Move frame input (ViECapturer) to webrtc/video/.
Renames ViECapturer to VideoCaptureInput and initializes several
parameters on construction instead of setters.

Also removes an old deadlock suppression.

BUG=1695, 2999
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53559004.

Cr-Commit-Position: refs/heads/master@{#9508}
2015-06-26 04:58:23 +00:00
Fredrik Solenberg
78fb3b3f8f C++11 in-class member initialization in Call configs.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1166263004.

Cr-Commit-Position: refs/heads/master@{#9416}
2015-06-11 10:38:46 +00:00
Miguel Casas-Sanchez
4765070b8d Rename I420VideoFrame to VideoFrame.
This is a mechanical change since it affects so many
files.
I420VideoFrame -> VideoFrame
and reformatted.

Rationale: in the next CL I420VideoFrame will
get an indication of Pixel Format (I420 for
starters) and of storage type: usually
UNOWNED, could be SHMEM, and in the near
future will be possibly TEXTURE. See
https://codereview.chromium.org/1154153003
for the change that happened in Cr.

BUG=4730, chromium:440843
R=jiayl@webrtc.org, niklas.enbom@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52629004

Cr-Commit-Position: refs/heads/master@{#9339}
2015-05-30 00:21:56 +00:00
perkj@webrtc.org
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
magjed@webrtc.org
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
perkj@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
32e8528581 Log configs when creating video streams in Call.
Adds VideoReceiveStream::Config::ToString and logs configs in both
Call::CreateVideoSendStream and Call::CreateVideoReceiverStream.

R=mflodman@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/41519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8075 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 10:09:39 +00:00
stefan@webrtc.org
742386a136 Enable payload-based padding by default and remove the API.
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 15:33:17 +00:00
pbos@webrtc.org
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
stefan@webrtc.org
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
pbos@webrtc.org
ad3b5a5c16 Move min transmit bitrate to VideoEncoderConfig.
min_transmit_bitrate_bps needs to be reconfigurable during a call (since
this is currently set only for screensharing through libjingle and can't
be set once and for all for the entire Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7518 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-24 09:23:21 +00:00
pbos@webrtc.org
bbe0a8517d Config struct for VideoEncoder.
Used for config parameters in common between multiple codecs as well as
the encoder-specific pointer. In particular this contains content mode
(realtime video vs. screenshare).

BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 12:30:25 +00:00
stefan@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
cb254aac3b Enable pacing by default and remove the option to disable it from the new API.
BUG=1672
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6416 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 15:12:25 +00:00
stefan@webrtc.org
fbb567dacd Add APIs to enable padding with redundant payloads.
Also makes a small change to the tests to remove flakiness. We can't do
BWE only based on rtp timestamps if we preemptively resend packets instead
of sending padding packets.

BUG=1812,2992
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6400 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 13:41:36 +00:00