Commit Graph

15 Commits

Author SHA1 Message Date
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
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
7b971e728b Remove extra_options from VideoCodec.
Constructing default options is racy when initializing multiple VP8
encoders in parallel. This is only used for VP8 temporal layers. Adding
TemporalLayerFactory to VP8 codec specifics instead of generic options.

Removes the last webrtc::Config uses/includes from video code.

Also removes VideoCodec equality operators which are no longer in use.

BUG=webrtc:5410
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1606613003 .

Cr-Commit-Position: refs/heads/master@{#11307}
2016-01-19 15:26:24 +00:00
13f61dfea5 Move fake-handle frame creation into test target.
Renames CreateFakeNativeHandleFrame to FakeNativeHandle::CreateFrame and
moves into test.gyp target 'fake_video_frames' which contains previous
frame_generator target.

Removes unused warnings from includers of
webrtc/test/fake_texture_frame.h which did not use the function above.

BUG=webrtc:5398
R=kjellander@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1554223002 .

Cr-Commit-Position: refs/heads/master@{#11149}
2016-01-04 21:36:49 +00:00
5811a39f14 Replace EventWrapper in video/, test/ and call/.
Makes use of rtc::Event which is simpler and can be used without
allocating additional objects on the heap.

Does not modify test/channel_transport/.

BUG=
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1487893004 .

Cr-Commit-Position: refs/heads/master@{#10968}
2015-12-10 12:03:00 +00:00
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
ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

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

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00
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
f4d23b2254 Remove MockVideoCapturer
This class is not used.

Review URL: https://codereview.webrtc.org/1403783002

Cr-Commit-Position: refs/heads/master@{#10360}
2015-10-21 16:56:14 +00:00
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
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
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
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
db0cf7624e Add test for dropping repeated NTP timestamps.
Regression test for enforcing that frames with repeated or old NTP
timestamps are dropped.

BUG=chromium:480953, webrtc:4615
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1220193002

Cr-Commit-Position: refs/heads/master@{#9533}
2015-07-02 11:14:50 +00:00
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