91 Commits

Author SHA1 Message Date
sprang
e5c4a810e2 Move RTP keep-alive config from VideoSendStream::Config to Call::Config
This makes more sense since logically it's a transport level feature,
not a media stream feature. Even if the implementation details forces it
to be an rtp stream detail, for the moment.

BUG=webrtc:7907

Review-Url: https://codereview.webrtc.org/2978503002
Cr-Commit-Position: refs/heads/master@{#18963}
2017-07-11 10:44:17 +00:00
Edward Lemur
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
sprang
168794c43c Implement RTP keepalive in native stack.
BUG=webrtc:7907

Review-Url: https://codereview.webrtc.org/2960363002
Cr-Commit-Position: refs/heads/master@{#18912}
2017-07-06 11:38:06 +00:00
Henrik Kjellander
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
kjellander
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
asapersson
3c81a1afd8 Add field trial for balanced degradation preference.
BUG=webrtc:7607

Review-Url: https://codereview.webrtc.org/2923563002
Cr-Commit-Position: refs/heads/master@{#18589}
2017-06-14 12:52:21 +00:00
asapersson
09f0561675 Update adaptation stats to support degradations in both resolution and framerate.
Add AdaptCounter class which holds the number of downgrade counts per degradation way (resolution/fps) and reason (cpu/quality).

BUG=webrtc:7607

Review-Url: https://codereview.webrtc.org/2871623002
Cr-Commit-Position: refs/heads/master@{#18156}
2017-05-16 06:40:18 +00:00
asapersson
fab67071a4 Add number of quality adapt changes to VideoSendStream stats.
The stats will be used in an upcoming cl to track the rate of adapt changes.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2775173004
Cr-Commit-Position: refs/heads/master@{#17528}
2017-04-04 12:51:49 +00:00
sprang
c5d62e29ca Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
Reason for revert:
Seem to be a flaky test rather than an issue with this cl. Creating reland, will add code to reduce flakiness to that test.

Original issue's description:
> Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
>
> Reason for revert:
> This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780
>
> Original issue's description:
> > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
> >
> > Reason for revert:
> > Found issue with test case, will add fix to reland cl.
> >
> > Original issue's description:
> > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
> > >
> > > Reason for revert:
> > > Breaks perf tests:
> > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
> > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325
> > >
> > > Original issue's description:
> > > > Add framerate to VideoSinkWants and ability to signal on overuse
> > > >
> > > > In ViEEncoder, try to reduce framerate instead of resolution if the
> > > > current degradation preference is maintain-resolution rather than
> > > > balanced.
> > > >
> > > > BUG=webrtc:4172
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2716643002
> > > > Cr-Commit-Position: refs/heads/master@{#17327}
> > > > Committed: 72acf25261
> > >
> > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:4172
> > >
> > > Review-Url: https://codereview.webrtc.org/2764133002
> > > Cr-Commit-Position: refs/heads/master@{#17331}
> > > Committed: 8b45b11144
> >
> > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:4172
> >
> > Review-Url: https://codereview.webrtc.org/2781433002
> > Cr-Commit-Position: refs/heads/master@{#17474}
> > Committed: 3ea3c77e93
>
> TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2783183003
> Cr-Commit-Position: refs/heads/master@{#17477}
> Committed: f9ed235c9b

R=ilnik@webrtc.org,stefan@webrtc.org
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2789823002
Cr-Commit-Position: refs/heads/master@{#17498}
2017-04-03 06:53:04 +00:00
lliuu
f9ed235c9b Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
Reason for revert:
This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780

Original issue's description:
> Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
>
> Reason for revert:
> Found issue with test case, will add fix to reland cl.
>
> Original issue's description:
> > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
> >
> > Reason for revert:
> > Breaks perf tests:
> > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
> > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325
> >
> > Original issue's description:
> > > Add framerate to VideoSinkWants and ability to signal on overuse
> > >
> > > In ViEEncoder, try to reduce framerate instead of resolution if the
> > > current degradation preference is maintain-resolution rather than
> > > balanced.
> > >
> > > BUG=webrtc:4172
> > >
> > > Review-Url: https://codereview.webrtc.org/2716643002
> > > Cr-Commit-Position: refs/heads/master@{#17327}
> > > Committed: 72acf25261
> >
> > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:4172
> >
> > Review-Url: https://codereview.webrtc.org/2764133002
> > Cr-Commit-Position: refs/heads/master@{#17331}
> > Committed: 8b45b11144
>
> TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2781433002
> Cr-Commit-Position: refs/heads/master@{#17474}
> Committed: 3ea3c77e93

TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2783183003
Cr-Commit-Position: refs/heads/master@{#17477}
2017-03-30 17:44:38 +00:00
sprang
3ea3c77e93 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
Reason for revert:
Found issue with test case, will add fix to reland cl.

Original issue's description:
> Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
>
> Reason for revert:
> Breaks perf tests:
> https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
> https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325
>
> Original issue's description:
> > Add framerate to VideoSinkWants and ability to signal on overuse
> >
> > In ViEEncoder, try to reduce framerate instead of resolution if the
> > current degradation preference is maintain-resolution rather than
> > balanced.
> >
> > BUG=webrtc:4172
> >
> > Review-Url: https://codereview.webrtc.org/2716643002
> > Cr-Commit-Position: refs/heads/master@{#17327}
> > Committed: 72acf25261
>
> TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2764133002
> Cr-Commit-Position: refs/heads/master@{#17331}
> Committed: 8b45b11144

TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2781433002
Cr-Commit-Position: refs/heads/master@{#17474}
2017-03-30 14:23:48 +00:00
skvlad
8b45b11144 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
Reason for revert:
Breaks perf tests:
https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325

Original issue's description:
> Add framerate to VideoSinkWants and ability to signal on overuse
>
> In ViEEncoder, try to reduce framerate instead of resolution if the
> current degradation preference is maintain-resolution rather than
> balanced.
>
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2716643002
> Cr-Commit-Position: refs/heads/master@{#17327}
> Committed: 72acf25261

TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2764133002
Cr-Commit-Position: refs/heads/master@{#17331}
2017-03-21 20:26:06 +00:00
sprang
72acf25261 Add framerate to VideoSinkWants and ability to signal on overuse
In ViEEncoder, try to reduce framerate instead of resolution if the
current degradation preference is maintain-resolution rather than
balanced.

BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2716643002
Cr-Commit-Position: refs/heads/master@{#17327}
2017-03-21 18:54:11 +00:00
brandtr
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
sergeyu
80ed35e21c Implement periodic bandwidth probing in application-limited region.
Now ProbeController can send periodic bandwidth probes when in
application-limited region. This will allow to maintain correct
bottleneck bandwidth estimate, even not all bandwidth is being used.
The feature is not enabled by default, but can be enabled with a flag.
Interval between probes is currently set to 5 seconds.

BUG=webrtc:6332

Review-Url: https://codereview.webrtc.org/2504023002
Cr-Commit-Position: refs/heads/master@{#15279}
2016-11-28 21:11:24 +00:00
aleloi
a8eb756a34 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.

Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.

transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.

NOTRY=True

BUG=webrtc:5589, webrtc:5878, webrtc:6785

Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
2016-11-28 15:02:19 +00:00
asapersson
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
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