Commit Graph

171 Commits

Author SHA1 Message Date
59cac99c9a Report minimum PSNR in VideoQualityTest and save corresponding frame to file
BUG=none

Review-Url: https://codereview.webrtc.org/2976373002
Cr-Commit-Position: refs/heads/master@{#19130}
2017-07-25 12:45:03 +00:00
9843695166 Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback
BUG=none

Review-Url: https://codereview.webrtc.org/2974903002
Cr-Commit-Position: refs/heads/master@{#18996}
2017-07-13 07:47:03 +00:00
863f03ba38 Fix video_replay tool to respect RTX stream and fix default parameters.
Defaults are consistent with these used in CallTest.

BUG=none

Review-Url: https://codereview.webrtc.org/2972423002
Cr-Commit-Position: refs/heads/master@{#18961}
2017-07-11 09:38:36 +00:00
6cc25614a9 Remove webrtc::VideoEncoderFactory
Replace the use of webrtc::VideoEncoderFactory with
cricket::WebRtcVideoEncoderFactory and remove the adapter classes
between these two factory types.

Some code changes were necessary in order to accomplish this:
 * Move SimulcastEncoderAdapter from
   webrtc/modules/video_coding/codecs/vp8 to webrtc/media/engine (that's
   where it's used).
 * Rename simulcast_unittest.h to simulcast_test_utility.h and make it
   into it's own target, because it's used from both
   simulcast_unittest.cc and simulcast_encoder_adapter_unittest.cc.
 * Remove ownership of the encoder factory from SimulcastEncoderAdapter,
   and make the necessary changes in surrounding code.

The goal with this CL is to clean up the code, and also to free up
the name webrtc::VideoEncoderFactory for future use.

BUG=webrtc:7925

Review-Url: https://codereview.webrtc.org/2964953002
Cr-Commit-Position: refs/heads/master@{#18945}
2017-07-10 10:26:36 +00:00
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
889d9654f7 Fix issue with zero rtt reports when using FlexFEC and add perf test.
BUG=webrtc:7938

Review-Url: https://codereview.webrtc.org/2966153002
Cr-Commit-Position: refs/heads/master@{#18898}
2017-07-05 10:03:02 +00:00
a80c16a67c Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
This reverts commit c3771cc4d37f5573fe53b7c7cff295a4f0f9560f.
(breaks downstream internal project)

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2972463002 .
Cr-Commit-Position: refs/heads/master@{#18873}
2017-07-01 14:48:18 +00:00
c3771cc4d3 Update includes for webrtc/{base => rtc_base} rename (2/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.

Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
2017-06-30 20:42:44 +00:00
cb8f045d9f Fix receiving FlexFEC in video_loopback.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2965503006
Cr-Commit-Position: refs/heads/master@{#18847}
2017-06-30 09:34:20 +00:00
a9cc40b7d2 Allow an external audio processing module to be used in WebRTC
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]

Allow an external audio processing module to be used in WebRTC

This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.

As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.

BUG=webrtc:7775

Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
2017-06-29 15:32:09 +00:00
267041c470 Fix deadlock in webrtc_perf_tests
Reenable hanging tests on Mac.

Deadlock happened because the following locks were grabbed by two threads at the end of a test:
Thread 1:
CapturedFrameForwarder::AddOrUpdateSink() locks CapturedFrameForwarder::crit_ and calls
FrameGeneratorCapturer::AddOrUpdateSink() what tries to lock FrameGeneratorCapturer::lock_.

Thread 2:
FrameGeneratorCapturer::InsertFrame() locks FrameGeneratorCapturer::lock_ and calls
CapturedFrameForwarder::OnFrame() which tries to lock CapturedFrameForwarder::crit_.

So two threads are locking two same locks in different orders which may cause deadlock.

BUG=webrtc:7870

Review-Url: https://codereview.webrtc.org/2955083002
Cr-Commit-Position: refs/heads/master@{#18783}
2017-06-27 14:21:01 +00:00
1168fd4ed5 What can't loopback test be more like full stack test?
It can; this CL makes it a lot closer, if not all the way to a merge.
Performance from video_loopback and screenshare_loopback should now
match what we're seeing in FullStackTest, which will make debugging and
assesment of quality differences much easier.

It also adds the ability to view all of the simulcast streams at once,
in separate windows.

BUG=webrtc:7694

Review-Url: https://codereview.webrtc.org/2946893003
Cr-Commit-Position: refs/heads/master@{#18703}
2017-06-21 16:00:17 +00:00
04f4d126f8 Implement timing frames.
Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}
2017-06-19 14:18:55 +00:00
6b826ef66d Add cropping to VIEEncoder to match simulcast streams resolution
Detect when simulcaststreamfactory adjust resolution and remeber cropping
parameters in VIEEncoder.
Expose EncoderStreamFactory in webrtcvideoengine2.

BUG=webrtc:7375, webrtc:6958

Review-Url: https://codereview.webrtc.org/2936393002
Cr-Commit-Position: refs/heads/master@{#18632}
2017-06-16 13:53:48 +00:00
67561a6411 Use the same QP max for tests as in production
BUG=webrtc:7664

Review-Url: https://codereview.webrtc.org/2941023002
Cr-Commit-Position: refs/heads/master@{#18611}
2017-06-15 13:34:42 +00:00
8e857d10fd Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device.
BUG=webrtc:7719

Change-Id: Iddc66188341c0c90e96766dff671ac3863bf3f5d
Reviewed-on: https://chromium-review.googlesource.com/517523
Commit-Queue: Peter Boström <pbos@webrtc.org>
Reviewed-by: Peter Boström <pbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18392}
2017-06-01 21:10:29 +00:00
48368ad6c6 Fixing video loopback test with encoder factory.
BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2870123002
Cr-Commit-Position: refs/heads/master@{#18079}
2017-05-10 11:06:11 +00:00
b82ac6aed3 Fix video_loopback to work with -duration flag: add missing ntp_timestamp to frames
BUG=none

Review-Url: https://codereview.webrtc.org/2852463002
Cr-Commit-Position: refs/heads/master@{#17967}
2017-05-02 07:48:41 +00:00
20a4b3fb2a Injectable audio encoders: WebRtcVoiceEngine and company
These are the changes made to WebRtcVoiceEngine and surrounding
code. It still contains some things that are inelegant, like how
AudioCodecSpec and AudioFormatInfo is ferried around in
SendCodecSpec. This should probably be resolved before landing.

There are also a few test still that are disabled. They should be
removed or fixed, as the case may be.

I've put this CL up to get a better overview of the changes made and
how reviewable they are.

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2705093002
Cr-Commit-Position: refs/heads/master@{#17904}
2017-04-27 09:08:52 +00:00
8d8185c774 Add command-line param to screenshare_loopback to specify a list of slides
BUG=none

Review-Url: https://codereview.webrtc.org/2814023003
Cr-Commit-Position: refs/heads/master@{#17670}
2017-04-12 11:52:55 +00:00
00d802b6ee Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
Reason for revert:
Fix failing bots.

BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2816493002
Cr-Commit-Position: refs/heads/master@{#17658}
2017-04-11 17:34:31 +00:00
27c46e2872 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ )
Reason for revert:
Breaks android buildbots.

Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with appropriate changes to API to not break depending projects.
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2812913002
> Cr-Commit-Position: refs/heads/master@{#17651}
> Committed: 774f6b4b96

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2809653004
Cr-Commit-Position: refs/heads/master@{#17653}
2017-04-11 13:20:05 +00:00
774f6b4b96 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
Reason for revert:
Reland with appropriate changes to API to not break depending projects.

Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2812913002
Cr-Commit-Position: refs/heads/master@{#17651}
2017-04-11 13:12:37 +00:00
29dbb1992a Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ )
Reason for revert:
Relanded by mistake.

Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with fixes which break API
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2811963002
> Cr-Commit-Position: refs/heads/master@{#17645}
> Committed: 4fa0c4f97f

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2810923004
Cr-Commit-Position: refs/heads/master@{#17648}
2017-04-11 11:49:07 +00:00
4fa0c4f97f Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
Reason for revert:
Reland with fixes which break API

Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2811963002
Cr-Commit-Position: refs/heads/master@{#17645}
2017-04-11 11:01:43 +00:00
5721866808 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
Reason for revert:
Breaks dependent projects.

Original issue's description:
> Add content type information to Encoded Images and add corresponding RTP extension header.
> Use it to separate UMA e2e delay metric between screenshare from video.
> Content type extension is set based on encoder settings and processed and decoders.
>
> Also,
> Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
>
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2772033002
> Cr-Commit-Position: refs/heads/master@{#17640}
> Committed: 64e739aeae

TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2816463002
Cr-Commit-Position: refs/heads/master@{#17644}
2017-04-11 10:59:43 +00:00
64e739aeae Add content type information to Encoded Images and add corresponding RTP extension header.
Use it to separate UMA e2e delay metric between screenshare from video.
Content type extension is set based on encoder settings and processed and decoders.

Also,
Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.

BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2772033002
Cr-Commit-Position: refs/heads/master@{#17640}
2017-04-11 08:46:04 +00:00
20c84ccd48 Making FakeNetworkPipe demux audio and video packets.
BUG=None

Review-Url: https://codereview.webrtc.org/2794243002
Cr-Commit-Position: refs/heads/master@{#17629}
2017-04-10 23:57:57 +00:00
4fb651dd22 Event log cleanup in tests.
TBR=stefan@webrtc.org
BUG=none

Review-Url: https://codereview.webrtc.org/2806723002
Cr-Commit-Position: refs/heads/master@{#17614}
2017-04-10 10:54:05 +00:00
2da7a24fcd Add back sender_time and receiver_time metrics to full stack tests.
BUG=none

Review-Url: https://codereview.webrtc.org/2781323002
Cr-Commit-Position: refs/heads/master@{#17462}
2017-03-30 08:02:15 +00:00
e5ad5ca06a Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
Reason for revert:
Intend to fix perf failures and reland.

Original issue's description:
> Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
>
> Reason for revert:
> Reverting since this seems to break multiple WebRTC Perf buildbots
>
> Original issue's description:
> > Don't hardcode MediaType::ANY in FakeNetworkPipe.
> >
> > Instead let each test set the appropriate media type. This simplifies
> > demuxing in Call and later in RtpTransportController.
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2774463003
> > Cr-Commit-Position: refs/heads/master@{#17418}
> > Committed: 9c47b00e24
>
> TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2784543002
> Cr-Commit-Position: refs/heads/master@{#17427}
> Committed: 3a3bd50610

TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2783853002
Cr-Commit-Position: refs/heads/master@{#17459}
2017-03-30 06:57:43 +00:00
3a3bd50610 Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
Reason for revert:
Reverting since this seems to break multiple WebRTC Perf buildbots

Original issue's description:
> Don't hardcode MediaType::ANY in FakeNetworkPipe.
>
> Instead let each test set the appropriate media type. This simplifies
> demuxing in Call and later in RtpTransportController.
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2774463003
> Cr-Commit-Position: refs/heads/master@{#17418}
> Committed: 9c47b00e24

TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2784543002
Cr-Commit-Position: refs/heads/master@{#17427}
2017-03-28 16:40:59 +00:00
9c47b00e24 Don't hardcode MediaType::ANY in FakeNetworkPipe.
Instead let each test set the appropriate media type. This simplifies
demuxing in Call and later in RtpTransportController.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2774463003
Cr-Commit-Position: refs/heads/master@{#17418}
2017-03-28 11:59:41 +00:00
4c8b9425b0 Adding audio DTX to video loopback test.
BUG=None

Review-Url: https://codereview.webrtc.org/2768473002
Cr-Commit-Position: refs/heads/master@{#17317}
2017-03-21 11:11:43 +00:00
6ef1b34aae Fix perf test regression for screenshare and vp9.
Turns out temporal_layer_thresholds_bps doesn't work quite as expected.
It's for instance not honored at all for normal VP8 video. We need to
take a pass over this in general.

BUG=chromium:700297

Review-Url: https://codereview.webrtc.org/2744823002
Cr-Commit-Position: refs/heads/master@{#17199}
2017-03-13 09:01:32 +00:00
ff2ebf5e30 Clean up perf metrics and report ramp-up stats for fewer tests.
BUG=None

Review-Url: https://codereview.webrtc.org/2738183004
Cr-Commit-Position: refs/heads/master@{#17197}
2017-03-13 08:27:03 +00:00
cb8c1467bd Add FullStack test for simulcast screenshare mode.
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2745523002
Cr-Commit-Position: refs/heads/master@{#17150}
2017-03-09 17:23:30 +00:00
f89a738626 Disable failing fullstack test with 15 thumbnail streams
BUG=webrtc:7301

Review-Url: https://codereview.webrtc.org/2739613003
Cr-Commit-Position: refs/heads/master@{#17095}
2017-03-07 14:15:27 +00:00
a014cc5eb1 Reland of "Added large room scenario to full-stack tests"
Added thumbnail streams functionality to video quality test.

Changed simulcast full-stack tests to be 30fps instead of 50 to
better reflect real usecases (expect all kind of perf metrics to
improve).

BUG=webrtc:7095, webrtc:7301

Review-Url: https://codereview.webrtc.org/2733943003
Cr-Commit-Position: refs/heads/master@{#17092}
2017-03-07 12:21:04 +00:00
bfb124596e Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ )
Reason for revert:
webrtc_perf_tests crashes on android and windows due to too large test.

Original issue's description:
> Added large room scenario to full-stack tests. Added thumbnail streams functionality to video quality test.
>
> Changed simulcast full-stack tests to be 30fps instead of 50 to better reflect real usecases (expect all kind of perf metrics to improve).
>
> BUG=webrtc:7095
>
> Review-Url: https://codereview.webrtc.org/2730073002
> Cr-Commit-Position: refs/heads/master@{#17068}
> Committed: d8bd1b1d82

TBR=sprang@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2734753004
Cr-Commit-Position: refs/heads/master@{#17071}
2017-03-06 15:35:13 +00:00
d8bd1b1d82 Added large room scenario to full-stack tests. Added thumbnail streams functionality to video quality test.
Changed simulcast full-stack tests to be 30fps instead of 50 to better reflect real usecases (expect all kind of perf metrics to improve).

BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2730073002
Cr-Commit-Position: refs/heads/master@{#17068}
2017-03-06 14:10:28 +00:00
9fd9f6c15f Fixed VP8 simulcast full-stack-tests to not decode non-selected streams.
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2730433005
Cr-Commit-Position: refs/heads/master@{#16976}
2017-03-02 16:10:10 +00:00
68af10de7f Revert of fixed VP8 simulcast to not decode non-selected streams (patchset #5 id:80001 of https://codereview.webrtc.org/2728553003/ )
Reason for revert:
Causes regression in VP8 simulcast metrics (receive time, encoded frame size, etc) as two excluded streams' decoders request keyframes periodically, which affects metrics of a selected stream.

Original issue's description:
> In full-stack tests: fixed VP8 simulcast to not decode non-selected streams.
>
> BUG=webrtc:7095
>
> Review-Url: https://codereview.webrtc.org/2728553003
> Cr-Commit-Position: refs/heads/master@{#16948}
> Committed: 8dccd67520

TBR=sprang@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2729623005
Cr-Commit-Position: refs/heads/master@{#16967}
2017-03-02 12:59:33 +00:00
8dccd67520 In full-stack tests: fixed VP8 simulcast to not decode non-selected streams.
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2728553003
Cr-Commit-Position: refs/heads/master@{#16948}
2017-03-01 15:10:30 +00:00
daa574d1db Adding memory usage metric to full-stack video tests (only for WIN until we find more stable method to measure memory usage)
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2719013004
Cr-Commit-Position: refs/heads/master@{#16945}
2017-03-01 14:46:05 +00:00
c1b57a15bf Test field trial group with startswith rather than equals.
BUG=webrtc:7266

Review-Url: https://codereview.webrtc.org/2717973005
Cr-Commit-Position: refs/heads/master@{#16915}
2017-02-28 16:50:47 +00:00
a8ba195db5 Replace test::FrameGenerator::ChromaGenerator with new FrameGenerator::SquareGenerator The problem with the ChromaGenerator is that the VP8 encoder produce a too low bitrate for each frame. The squaregenerator make the VP8 encoder produce about 600kbit/s at VGA.
SquareGenerator is a FrameGenerator that draws 10 randomly sized and colored
squares. Between each new generated frame, the squares are moved slightly
towards the lower right corner.

BUG=webrtc:7192

Review-Url: https://codereview.webrtc.org/2705973002
Cr-Commit-Position: refs/heads/master@{#16870}
2017-02-27 14:52:10 +00:00
1e7732c3d9 Fixed Full stack tests to correctly process selected TL and SL while
calculating frame sizes. Added actual_bitrate metric which also accounts
for TL and SL info. Metric encoded_frame_size calculation is cleaned up. Perf alerts should be ignored.

BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2709483009
Cr-Commit-Position: refs/heads/master@{#16800}
2017-02-23 13:07:56 +00:00
df92c5cb8c Adding cpu measurments to video_quality_tests
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2711493002
Cr-Commit-Position: refs/heads/master@{#16791}
2017-02-23 10:08:44 +00:00
0f8b403eb5 Introduce a new constructor to PlatformThread.
The new constructor introduces two new changes:

* Support specifying thread priority at construction time.
  - Moving forward, the SetPriority() method will be removed.
* New thread function type.
  - The new type has 'void' as a return type and a polling loop
    inside PlatformThread, is not used.

The old function type is still supported until all places have been moved over.

In this CL, the first steps towards deprecating the old mechanism are taken
by moving parts of the code that were simple to move, over to the new callback
type.

BUG=webrtc:7187

Review-Url: https://codereview.webrtc.org/2708723003
Cr-Commit-Position: refs/heads/master@{#16779}
2017-02-22 19:22:05 +00:00