This CL brings us one step closer to removing CodecDatabase and
GenericEncoder, by removing the static VCM::Codec(). Codec specific
methods are moved to video_encoder.cc (they already belonged to this
class) and getting default generic codec settings has been moved to a
test specific file.
This CL also makes video_encoder.h pass style guide and lint checks,
since these checks are triggered with the new video_encoder.cc file.
BUG=webrtc:8064
Review-Url: https://codereview.webrtc.org/2993923002
Cr-Commit-Position: refs/heads/master@{#19303}
Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single
timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported.
The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process).
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2946413002
Cr-Commit-Position: refs/heads/master@{#18909}
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}
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: 774f6b4b96TBR=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}
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: 5721866808TBR=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}
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: 4fa0c4f97fTBR=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}
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: 5721866808TBR=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}
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: 64e739aeaeTBR=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}
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}
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}
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}
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: 04926b8264TBR=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}
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: f20dd0014dTBR=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}
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: c08c191f7dTBR=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}
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: 0f0763d86dTBR=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}
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}
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.
BUG=webrtc:5880
Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
When the target bitrate is zero, currently VideoSendStream.Stats.target_media_bitrate_bps show the last set rate before the target was set to zero.
BUG=webrtc::5687 b/29574845
Review-Url: https://codereview.webrtc.org/2122743003
Cr-Commit-Position: refs/heads/master@{#13386}
And move encoder name cb to VCMSendStatisticsCallback.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1900193004
Cr-Commit-Position: refs/heads/master@{#12596}
Framerate-reduction code is disabled on all platforms, and this code
adds complexity. It's necessary to react fast, especially on mobile
platforms or other bad network conditions and framerate reduction adds
another step between HD and QVGA.
BUG=webrtc:5678, webrtc:5830
R=jackychen@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1885893002 .
Cr-Commit-Position: refs/heads/master@{#12503}
Reason for revert:
A fix is being prepared downstream so this can now go in.
Original issue's description:
> Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ )
>
> Reason for revert:
> API changes broke downstream.
>
> Original issue's description:
> > Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
> > EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
> > EncodedImageCallback can of course be cleaned up in the future.
> >
> > This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
> >
> > BUG=webrtc::5687
> >
> > Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb
> > Cr-Commit-Position: refs/heads/master@{#12436}
>
> TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5687
>
> Committed: https://crrev.com/a261e6136655af33f283eda8e60a6dd93dd746a4
> Cr-Commit-Position: refs/heads/master@{#12441}
TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@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/1905583002
Cr-Commit-Position: refs/heads/master@{#12442}
Reason for revert:
API changes broke downstream.
Original issue's description:
> Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
> EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
> EncodedImageCallback can of course be cleaned up in the future.
>
> This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
>
> BUG=webrtc::5687
>
> Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb
> Cr-Commit-Position: refs/heads/master@{#12436}
TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@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/1903193002
Cr-Commit-Position: refs/heads/master@{#12441}
EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
EncodedImageCallback can of course be cleaned up in the future.
This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
BUG=webrtc::5687
Review URL: https://codereview.webrtc.org/1897233002
Cr-Commit-Position: refs/heads/master@{#12436}
Things done/implemented in this CL:
- An interface that can send Nack (VCMNackSender).
- An interface that can request KeyFrames (VCMKeyFrameRequestSender).
- The nack module (NackModule).
- A set of convenience functions for modular numbers (mod_ops.h).
BUG=webrtc:5514
Review URL: https://codereview.webrtc.org/1715673002
Cr-Commit-Position: refs/heads/master@{#11882}
This argument is never used as a reference and the pointer that's bound
to the const reference may be nullptr. This is undefined behavior and
barks under UBSan.
BUG=webrtc:5124
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1642863003 .
Cr-Commit-Position: refs/heads/master@{#11418}
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.
To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417283007 .
Cr-Commit-Position: refs/heads/master@{#10694}
