Commit Graph

26 Commits

Author SHA1 Message Date
b1db3702f7 Delete unsupported method VideoCodingModule::RegisterDecoderTimingCallback.
The implementation behind this method has been a noop for a long time.

BUG=none

Review-Url: https://codereview.webrtc.org/2757843002
Cr-Commit-Position: refs/heads/master@{#17286}
2017-03-17 12:35:43 +00:00
a45102f7b4 Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )
Reason for revert:
Fix here: https://codereview.chromium.org/2708593003

Original issue's description:
> Revert Make the new jitter buffer the default jitter buffer.
>
> Speculative revert of https://codereview.chromium.org/2656983002/ to see if it fixes a downstream bug.
>
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2682073003
> Cr-Commit-Position: refs/heads/master@{#16492}
> Committed: e525d6aba6

TBR=nisse@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2704183002
Cr-Commit-Position: refs/heads/master@{#16772}
2017-02-22 13:30:39 +00:00
cc452e1179 Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
Reason for revert:
Fix the problem.

Original issue's description:
> Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
>
> Reason for revert:
> Breaks downstream build.
>
> Original issue's description:
> > Add QP sum stats for received streams.
> >
> > This is not implemented yet in any of the decoders.
> >
> > BUG=webrtc:6541
> >
> > Review-Url: https://codereview.webrtc.org/2649133005
> > Cr-Commit-Position: refs/heads/master@{#16475}
> > Committed: ff0e72fd16
>
> TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,sakal@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6541
>
> Review-Url: https://codereview.webrtc.org/2680893002 .
> Cr-Commit-Position: refs/heads/master@{#16480}
> Committed: 69fb2cca4d

TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2681663005
Cr-Commit-Position: refs/heads/master@{#16511}
2017-02-09 12:53:45 +00:00
e525d6aba6 Revert Make the new jitter buffer the default jitter buffer.
Speculative revert of https://codereview.chromium.org/2656983002/ to see if it fixes a downstream bug.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2682073003
Cr-Commit-Position: refs/heads/master@{#16492}
2017-02-08 13:25:42 +00:00
69fb2cca4d Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
Reason for revert:
Breaks downstream build.

Original issue's description:
> Add QP sum stats for received streams.
>
> This is not implemented yet in any of the decoders.
>
> BUG=webrtc:6541
>
> Review-Url: https://codereview.webrtc.org/2649133005
> Cr-Commit-Position: refs/heads/master@{#16475}
> Committed: ff0e72fd16

TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2680893002 .
Cr-Commit-Position: refs/heads/master@{#16480}
2017-02-07 18:59:25 +00:00
76bc8e858f Delete VideoReceiveStream::Config::pre_render_callback.
Also delete the class I420FrameCallback.

BUG=webrtc:7124

Review-Url: https://codereview.webrtc.org/2678343002
Cr-Commit-Position: refs/heads/master@{#16478}
2017-02-07 17:37:41 +00:00
ff0e72fd16 Add QP sum stats for received streams.
This is not implemented yet in any of the decoders.

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2649133005
Cr-Commit-Position: refs/heads/master@{#16475}
2017-02-07 15:15:17 +00:00
e5bd70223d Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ )
Reason for revert:
Incoming fix: https://codereview.chromium.org/2675693002/

Original issue's description:
> Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:290001 of https://codereview.chromium.org/2652043005/ )
>
> Reason for revert:
> Breaks downstream bots
>
> Original issue's description:
> > Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2638423003/ )
> >
> > Reason for revert:
> > Bugfixes related to the new jitter buffer has landed.
> >
> > Original issue's description:
> > > Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
> > >
> > > Reason for revert:
> > > Breaks tests downstream.
> > >
> > > Original issue's description:
> > > > Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
> > > >
> > > > Reason for revert:
> > > > Fix in this CL: https://codereview.chromium.org/2640793003/
> > > >
> > > > Original issue's description:
> > > > > Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
> > > > >
> > > > > Reason for revert:
> > > > > Breaks android bots.
> > > > >
> > > > > Original issue's description:
> > > > > > Make the new jitter buffer the default jitter buffer.
> > > > > >
> > > > > > This CL contains only the changes necessary to make the switch to the new jitter
> > > > > > buffer, clean up will be done in follow up CLs.
> > > > > >
> > > > > > In this CL:
> > > > > >  - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> > > > > >    new video jitter buffer the default one.
> > > > > >  - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> > > > > >    WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> > > > > >
> > > > > > BUG=webrtc:5514
> > > > > >
> > > > > > Review-Url: https://codereview.webrtc.org/2627463004
> > > > > > Cr-Commit-Position: refs/heads/master@{#16114}
> > > > > > Committed: 0f0763d86d
> > > > >
> > > > > TBR=stefan@webrtc.org,terelius@webrtc.org
> > > > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > > > NOPRESUBMIT=true
> > > > > NOTREECHECKS=true
> > > > > NOTRY=true
> > > > > BUG=webrtc:5514
> > > > >
> > > > > Review-Url: https://codereview.webrtc.org/2632123005
> > > > > Cr-Commit-Position: refs/heads/master@{#16117}
> > > > > Committed: c08c191f7d
> > > >
> > > > TBR=stefan@webrtc.org,terelius@webrtc.org
> > > > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > > > BUG=webrtc:5514
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2642753002
> > > > Cr-Commit-Position: refs/heads/master@{#16149}
> > > > Committed: f20dd0014d
> > >
> > > TBR=stefan@webrtc.org,terelius@webrtc.org,philipel@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:5514
> > >
> > > Review-Url: https://codereview.webrtc.org/2638423003
> > > Cr-Commit-Position: refs/heads/master@{#16159}
> > > Committed: 04926b8264
> >
> > TBR=stefan@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:5514
> >
> > Review-Url: https://codereview.webrtc.org/2652043005
> > Cr-Commit-Position: refs/heads/master@{#16293}
> > Committed: 09d6ef00fc
>
> TBR=stefan@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2656983002
> Cr-Commit-Position: refs/heads/master@{#16316}
> Committed: 27378f39ce

TBR=stefan@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2670183002
Cr-Commit-Position: refs/heads/master@{#16420}
2017-02-02 17:53:00 +00:00
27378f39ce Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:290001 of https://codereview.chromium.org/2652043005/ )
Reason for revert:
Breaks downstream bots

Original issue's description:
> Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2638423003/ )
>
> Reason for revert:
> Bugfixes related to the new jitter buffer has landed.
>
> Original issue's description:
> > Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
> >
> > Reason for revert:
> > Breaks tests downstream.
> >
> > Original issue's description:
> > > Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
> > >
> > > Reason for revert:
> > > Fix in this CL: https://codereview.chromium.org/2640793003/
> > >
> > > Original issue's description:
> > > > Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
> > > >
> > > > Reason for revert:
> > > > Breaks android bots.
> > > >
> > > > Original issue's description:
> > > > > Make the new jitter buffer the default jitter buffer.
> > > > >
> > > > > This CL contains only the changes necessary to make the switch to the new jitter
> > > > > buffer, clean up will be done in follow up CLs.
> > > > >
> > > > > In this CL:
> > > > >  - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> > > > >    new video jitter buffer the default one.
> > > > >  - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> > > > >    WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> > > > >
> > > > > BUG=webrtc:5514
> > > > >
> > > > > Review-Url: https://codereview.webrtc.org/2627463004
> > > > > Cr-Commit-Position: refs/heads/master@{#16114}
> > > > > Committed: 0f0763d86d
> > > >
> > > > TBR=stefan@webrtc.org,terelius@webrtc.org
> > > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > > NOPRESUBMIT=true
> > > > NOTREECHECKS=true
> > > > NOTRY=true
> > > > BUG=webrtc:5514
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2632123005
> > > > Cr-Commit-Position: refs/heads/master@{#16117}
> > > > Committed: c08c191f7d
> > >
> > > TBR=stefan@webrtc.org,terelius@webrtc.org
> > > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > > BUG=webrtc:5514
> > >
> > > Review-Url: https://codereview.webrtc.org/2642753002
> > > Cr-Commit-Position: refs/heads/master@{#16149}
> > > Committed: f20dd0014d
> >
> > TBR=stefan@webrtc.org,terelius@webrtc.org,philipel@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:5514
> >
> > Review-Url: https://codereview.webrtc.org/2638423003
> > Cr-Commit-Position: refs/heads/master@{#16159}
> > Committed: 04926b8264
>
> TBR=stefan@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2652043005
> Cr-Commit-Position: refs/heads/master@{#16293}
> Committed: 09d6ef00fc

TBR=stefan@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2656983002
Cr-Commit-Position: refs/heads/master@{#16316}
2017-01-27 10:19:05 +00:00
09d6ef00fc Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2638423003/ )
Reason for revert:
Bugfixes related to the new jitter buffer has landed.

Original issue's description:
> Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
>
> Reason for revert:
> Breaks tests downstream.
>
> Original issue's description:
> > Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
> >
> > Reason for revert:
> > Fix in this CL: https://codereview.chromium.org/2640793003/
> >
> > Original issue's description:
> > > Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
> > >
> > > Reason for revert:
> > > Breaks android bots.
> > >
> > > Original issue's description:
> > > > Make the new jitter buffer the default jitter buffer.
> > > >
> > > > This CL contains only the changes necessary to make the switch to the new jitter
> > > > buffer, clean up will be done in follow up CLs.
> > > >
> > > > In this CL:
> > > >  - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> > > >    new video jitter buffer the default one.
> > > >  - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> > > >    WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> > > >
> > > > BUG=webrtc:5514
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2627463004
> > > > Cr-Commit-Position: refs/heads/master@{#16114}
> > > > Committed: 0f0763d86d
> > >
> > > TBR=stefan@webrtc.org,terelius@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:5514
> > >
> > > Review-Url: https://codereview.webrtc.org/2632123005
> > > Cr-Commit-Position: refs/heads/master@{#16117}
> > > Committed: c08c191f7d
> >
> > TBR=stefan@webrtc.org,terelius@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:5514
> >
> > Review-Url: https://codereview.webrtc.org/2642753002
> > Cr-Commit-Position: refs/heads/master@{#16149}
> > Committed: f20dd0014d
>
> TBR=stefan@webrtc.org,terelius@webrtc.org,philipel@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2638423003
> Cr-Commit-Position: refs/heads/master@{#16159}
> Committed: 04926b8264

TBR=stefan@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2652043005
Cr-Commit-Position: refs/heads/master@{#16293}
2017-01-26 10:59:33 +00:00
04926b8264 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
Reason for revert:
Breaks tests downstream.

Original issue's description:
> Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
>
> Reason for revert:
> Fix in this CL: https://codereview.chromium.org/2640793003/
>
> Original issue's description:
> > Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
> >
> > Reason for revert:
> > Breaks android bots.
> >
> > Original issue's description:
> > > Make the new jitter buffer the default jitter buffer.
> > >
> > > This CL contains only the changes necessary to make the switch to the new jitter
> > > buffer, clean up will be done in follow up CLs.
> > >
> > > In this CL:
> > >  - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> > >    new video jitter buffer the default one.
> > >  - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> > >    WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> > >
> > > BUG=webrtc:5514
> > >
> > > Review-Url: https://codereview.webrtc.org/2627463004
> > > Cr-Commit-Position: refs/heads/master@{#16114}
> > > Committed: 0f0763d86d
> >
> > TBR=stefan@webrtc.org,terelius@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:5514
> >
> > Review-Url: https://codereview.webrtc.org/2632123005
> > Cr-Commit-Position: refs/heads/master@{#16117}
> > Committed: c08c191f7d
>
> TBR=stefan@webrtc.org,terelius@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2642753002
> Cr-Commit-Position: refs/heads/master@{#16149}
> Committed: f20dd0014d

TBR=stefan@webrtc.org,terelius@webrtc.org,philipel@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2638423003
Cr-Commit-Position: refs/heads/master@{#16159}
2017-01-19 08:06:17 +00:00
f20dd0014d Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
Reason for revert:
Fix in this CL: https://codereview.chromium.org/2640793003/

Original issue's description:
> Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
>
> Reason for revert:
> Breaks android bots.
>
> Original issue's description:
> > Make the new jitter buffer the default jitter buffer.
> >
> > This CL contains only the changes necessary to make the switch to the new jitter
> > buffer, clean up will be done in follow up CLs.
> >
> > In this CL:
> >  - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> >    new video jitter buffer the default one.
> >  - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> >    WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> >
> > BUG=webrtc:5514
> >
> > Review-Url: https://codereview.webrtc.org/2627463004
> > Cr-Commit-Position: refs/heads/master@{#16114}
> > Committed: 0f0763d86d
>
> TBR=stefan@webrtc.org,terelius@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2632123005
> Cr-Commit-Position: refs/heads/master@{#16117}
> Committed: c08c191f7d

TBR=stefan@webrtc.org,terelius@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2642753002
Cr-Commit-Position: refs/heads/master@{#16149}
2017-01-18 15:15:37 +00:00
c08c191f7d Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
Reason for revert:
Breaks android bots.

Original issue's description:
> Make the new jitter buffer the default jitter buffer.
>
> This CL contains only the changes necessary to make the switch to the new jitter
> buffer, clean up will be done in follow up CLs.
>
> In this CL:
>  - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
>    new video jitter buffer the default one.
>  - Moved WebRTC.Video.KeyFramesReceivedInPermille and
>    WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
>
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2627463004
> Cr-Commit-Position: refs/heads/master@{#16114}
> Committed: 0f0763d86d

TBR=stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2632123005
Cr-Commit-Position: refs/heads/master@{#16117}
2017-01-17 12:03:53 +00:00
0f0763d86d Make the new jitter buffer the default jitter buffer.
This CL contains only the changes necessary to make the switch to the new jitter
buffer, clean up will be done in follow up CLs.

In this CL:
 - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
   new video jitter buffer the default one.
 - Moved WebRTC.Video.KeyFramesReceivedInPermille and
   WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2627463004
Cr-Commit-Position: refs/heads/master@{#16114}
2017-01-17 11:31:15 +00:00
88499ecaca Moving/renaming webrtc/common.h.
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.

- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.

BUG=webrtc:5879

Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
2016-09-07 14:34:45 +00:00
4cd2790f17 Move RTP for synchroninzation and rename classes, files and variables.
This CL removes (almost) the last RTP references in VideoReceiveStream.
There are still references to RTPFragmentationHeader and SSRCs, which
will be dealt with later.

There are also new GUARDED_BY and thred checker added to the
synchronization class.

When there are othre transports than RTP, there will instead be an
interface + inheritance for RtpStreamReceiver and
RtpStreamSynchronizattion in VideoReceiveStream. This work will be done
when we actually know how we want to make thee transport interface.

BUG=webrtc:5838

Review-Url: https://codereview.webrtc.org/2216533002
Cr-Commit-Position: refs/heads/master@{#13655}
2016-08-05 13:28:50 +00:00
2e82f3821f Reland of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #1 id:1 of https://codereview.webrtc.org/2084873002/ )
Reason for revert:
Reverting the revert.  This change is not related to the failure on the Windows FYI bots.  The cause of the failure has been reverted in Chromium:
https://codereview.chromium.org/2081653004/

Original issue's description:
> Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #5 id:340001 of https://codereview.webrtc.org/2078873002/ )
>
> Reason for revert:
> Breaks chromium.webrtc.fyi
>
> https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win7%20Tester/builds/4719
> https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3120
>
> Original issue's description:
> > Reland of IncomingVideoStream refactoring.
> > This reland does not contain the non-smoothing part of the original implementation.  Instead, when smoothing is turned off, frame callbacks run on the decoder thread, as they did before.  This code path is used in Chrome.  As far as Chrome goes, the difference now is that there won't be an instance of IncomingVideoStream in between the decoder and the callback (i.e. fewer locks).  Other than that, no change for Chrome.
> >
> > Original issue's description (with non-smoothing references removed):
> >
> > Split IncomingVideoStream into two implementations, with smoothing and without.
> >
> > * Added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 6 locks.
> >
> > * Removed the Start/Stop methods from the IncomingVideoStream implementations.  Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running".  This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
> >
> > * Changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface.  This means that any implementation of that interface can be used and the decoder can be made to  just use the 'renderer' from the config.  Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing.  The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
> >
> > * The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
> >
> > * Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
> >
> > * Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes.  The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
> >
> > * Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction.  This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
> >
> > * Made the render delay value in VideoRenderFrames, const.
> >
> > BUG=chromium:620232
> > R=mflodman@webrtc.org, nisse@webrtc.org
> >
> > Committed: https://crrev.com/884c336c345d988974c2a69cea402b0fb8b07a63
> > Cr-Commit-Position: refs/heads/master@{#13219}
>
> TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:620232
>
> Committed: https://crrev.com/a536bfe70de38fe877245317a7f0b00bcf69cbd0
> Cr-Commit-Position: refs/heads/master@{#13229}

TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:620232

Review-Url: https://codereview.webrtc.org/2089613002
Cr-Commit-Position: refs/heads/master@{#13230}
2016-06-21 07:26:48 +00:00
a536bfe70d Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #5 id:340001 of https://codereview.webrtc.org/2078873002/ )
Reason for revert:
Breaks chromium.webrtc.fyi

https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win7%20Tester/builds/4719
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3120

Original issue's description:
> Reland of IncomingVideoStream refactoring.
> This reland does not contain the non-smoothing part of the original implementation.  Instead, when smoothing is turned off, frame callbacks run on the decoder thread, as they did before.  This code path is used in Chrome.  As far as Chrome goes, the difference now is that there won't be an instance of IncomingVideoStream in between the decoder and the callback (i.e. fewer locks).  Other than that, no change for Chrome.
>
> Original issue's description (with non-smoothing references removed):
>
> Split IncomingVideoStream into two implementations, with smoothing and without.
>
> * Added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 6 locks.
>
> * Removed the Start/Stop methods from the IncomingVideoStream implementations.  Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running".  This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
>
> * Changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface.  This means that any implementation of that interface can be used and the decoder can be made to  just use the 'renderer' from the config.  Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing.  The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
>
> * The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
>
> * Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
>
> * Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes.  The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
>
> * Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction.  This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
>
> * Made the render delay value in VideoRenderFrames, const.
>
> BUG=chromium:620232
> R=mflodman@webrtc.org, nisse@webrtc.org
>
> Committed: https://crrev.com/884c336c345d988974c2a69cea402b0fb8b07a63
> Cr-Commit-Position: refs/heads/master@{#13219}

TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:620232

Review-Url: https://codereview.webrtc.org/2084873002
Cr-Commit-Position: refs/heads/master@{#13229}
2016-06-21 07:08:58 +00:00
884c336c34 Reland of IncomingVideoStream refactoring.
This reland does not contain the non-smoothing part of the original implementation.  Instead, when smoothing is turned off, frame callbacks run on the decoder thread, as they did before.  This code path is used in Chrome.  As far as Chrome goes, the difference now is that there won't be an instance of IncomingVideoStream in between the decoder and the callback (i.e. fewer locks).  Other than that, no change for Chrome.

Original issue's description (with non-smoothing references removed):

Split IncomingVideoStream into two implementations, with smoothing and without.

* Added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 6 locks.

* Removed the Start/Stop methods from the IncomingVideoStream implementations.  Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running".  This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.

* Changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface.  This means that any implementation of that interface can be used and the decoder can be made to  just use the 'renderer' from the config.  Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing.  The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).

* The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.

* Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)

* Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes.  The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.

* Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction.  This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.

* Made the render delay value in VideoRenderFrames, const.

BUG=chromium:620232
R=mflodman@webrtc.org, nisse@webrtc.org

Review URL: https://codereview.webrtc.org/2078873002 .

Cr-Commit-Position: refs/heads/master@{#13219}
2016-06-20 17:43:10 +00:00
8e8222d0d2 Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #4 id:290001 of https://codereview.webrtc.org/2071473002/ )
Reason for revert:
Reverting again.  The perf regression does not seem to be related to dropping frames.

Original issue's description:
> Reland of Split IncomingVideoStream into two implementations, with smoothing and without.
>
> Original issue's description:
>
> Split IncomingVideoStream into two implementations, with smoothing and without.
>
> This CL fixes an issue with the non-smoothing implementation where frames were delivered on the decoder thread.  No-smoothing is now done in a separate class that uses a TaskQueue.  The implementation may drop frames if the renderer doesn't keep up and it doesn't block the decoder thread.
>
> Further work done:
>
> * I added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 5 locks.
>
> * I removed the Start/Stop methods from the IncomingVideoStream implementations.  Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running".  This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
>
> * I changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface.  This means that any implementation of that interface can be used and the decoder can be made to  just use the 'renderer' from the config.  Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing.  The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
>
> * The non-smoothing IncomingVideoStream implementation currently allows only 1 outstanding pending frame.  If we exceed that, the current frame won't be delivered to the renderer and instead we deliver the next one (since when this happens, the renderer is falling behind).
>
> * The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
>
> * Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
>
> * Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes.  The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
>
> * Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction.  This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
>
> * Made the render delay value in VideoRenderFrames, const.
>
> BUG=chromium:620232
> TBR=mflodman
>
> Committed: https://crrev.com/e03f8787377bbc03a4e00184bb14b7561b108cbb
> Cr-Commit-Position: refs/heads/master@{#13175}

TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:620232

Review-Url: https://codereview.webrtc.org/2071093002
Cr-Commit-Position: refs/heads/master@{#13176}
2016-06-16 22:44:11 +00:00
e03f878737 Reland of Split IncomingVideoStream into two implementations, with smoothing and without.
Original issue's description:

Split IncomingVideoStream into two implementations, with smoothing and without.

This CL fixes an issue with the non-smoothing implementation where frames were delivered on the decoder thread.  No-smoothing is now done in a separate class that uses a TaskQueue.  The implementation may drop frames if the renderer doesn't keep up and it doesn't block the decoder thread.

Further work done:

* I added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 5 locks.

* I removed the Start/Stop methods from the IncomingVideoStream implementations.  Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running".  This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.

* I changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface.  This means that any implementation of that interface can be used and the decoder can be made to  just use the 'renderer' from the config.  Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing.  The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).

* The non-smoothing IncomingVideoStream implementation currently allows only 1 outstanding pending frame.  If we exceed that, the current frame won't be delivered to the renderer and instead we deliver the next one (since when this happens, the renderer is falling behind).

* The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.

* Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)

* Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes.  The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.

* Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction.  This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.

* Made the render delay value in VideoRenderFrames, const.

BUG=chromium:620232
TBR=mflodman

Review-Url: https://codereview.webrtc.org/2071473002
Cr-Commit-Position: refs/heads/master@{#13175}
2016-06-16 20:29:12 +00:00
17c3cddf9d Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #23 id:430001 of https://codereview.webrtc.org/2035173002/ )
Reason for revert:
Reverting while we track down the issue on the Win10 bot.

Original issue's description:
> Split IncomingVideoStream into two implementations, with smoothing and without.
>
> This CL fixes an issue with the non-smoothing implementation where frames were delivered on the decoder thread.  No-smoothing is now done in a separate class that uses a TaskQueue.  The implementation may drop frames if the renderer doesn't keep up and it doesn't block the decoder thread.
>
> Further work done:
>
> * I added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 5 locks.
>
> * I removed the Start/Stop methods from the IncomingVideoStream implementations.  Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running".  This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
>
> * I changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface.  This means that any implementation of that interface can be used and the decoder can be made to  just use the 'renderer' from the config.  Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing.  The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
>
> * The non-smoothing IncomingVideoStream implementation currently allows only 1 outstanding pending frame.  If we exceed that, the current frame won't be delivered to the renderer and instead we deliver the next one (since when this happens, the renderer is falling behind).
>
> * The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
>
> * Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
>
> * Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes.  The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
>
> * Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction.  This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
>
> * Made the render delay value in VideoRenderFrames, const.
>
> BUG=
>
> Committed: https://crrev.com/1c7eef652b0aa22d8ebb0bfe2b547094a794be22
> Cr-Commit-Position: refs/heads/master@{#13129}

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

Review-Url: https://codereview.webrtc.org/2061363002
Cr-Commit-Position: refs/heads/master@{#13146}
2016-06-14 23:04:48 +00:00
1c7eef652b Split IncomingVideoStream into two implementations, with smoothing and without.
This CL fixes an issue with the non-smoothing implementation where frames were delivered on the decoder thread.  No-smoothing is now done in a separate class that uses a TaskQueue.  The implementation may drop frames if the renderer doesn't keep up and it doesn't block the decoder thread.

Further work done:

* I added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 5 locks.

* I removed the Start/Stop methods from the IncomingVideoStream implementations.  Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running".  This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.

* I changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface.  This means that any implementation of that interface can be used and the decoder can be made to  just use the 'renderer' from the config.  Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing.  The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).

* The non-smoothing IncomingVideoStream implementation currently allows only 1 outstanding pending frame.  If we exceed that, the current frame won't be delivered to the renderer and instead we deliver the next one (since when this happens, the renderer is falling behind).

* The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.

* Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)

* Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes.  The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.

* Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction.  This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.

* Made the render delay value in VideoRenderFrames, const.

BUG=

Review-Url: https://codereview.webrtc.org/2035173002
Cr-Commit-Position: refs/heads/master@{#13129}
2016-06-14 11:38:43 +00:00
d28db7fd65 Delete all use of tick_util.h.
Depends on Chrome cl https://codereview.chromium.org/1888003002/, which was landed some time ago.

BUG=webrtc:5740
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1888593004 .

Cr-Commit-Position: refs/heads/master@{#12674}
2016-05-10 14:31:58 +00:00
dc7d0d2ef0 Move, almost, all receive side references to RTP to RtpStreamReceiver.
There are still a few places in VideoReceiveStream where the RTP module
is explicitly used, e.g. setting up a/v sync, but it's a bigger task to
change and that will be done in a follow up instead of in this CL.

BUG=webrtc:5838

Review-Url: https://codereview.webrtc.org/1947913002
Cr-Commit-Position: refs/heads/master@{#12642}
2016-05-06 12:32:30 +00:00
cfc8e3b9ef Removed all RTP dependencies from ViEChannel and renamed class.
ViEChannel is now called VideoStreamReceiver.

There will be a follow up CL removing all rtp references from VideoReceiveStream, but that made this CL to big and it will be done separately.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/1929313002
Cr-Commit-Position: refs/heads/master@{#12619}
2016-05-04 04:22:12 +00:00