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}
Reason for revert:
Failures on the Linux Memcheck bot
Original issue's description:
> This approach passes packetization mode to the encoder as part of
> a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
>
> BUG=600254
>
> Committed: https://crrev.com/e59647b991f61cf1cf61b020356705e6c0f81257
> Cr-Commit-Position: refs/heads/master@{#15437}
TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=600254
Review-Url: https://codereview.webrtc.org/2558453002
Cr-Commit-Position: refs/heads/master@{#15441}
a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
BUG=600254
Review-Url: https://codereview.webrtc.org/2528343002
Cr-Commit-Position: refs/heads/master@{#15437}
This is yet another reland of https://codereview.webrtc.org/2434073003/
including two fixes:
1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that.
2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams.
Please review only the changes after patch set 1.
Original description:
Extract bitrate allocation of spatial/temporal layers out of codec impl.
This CL makes a number of intervowen changes:
* Add BitrateAllocation struct, that contains a codec independent view
of how the target bitrate is distributed over spatial and temporal
layers.
* Adds the BitrateAllocator interface, which takes a bitrate and frame
rate and produces a BitrateAllocation.
* A default (non layered) implementation is added, and
SimulcastRateAllocator is extended to fully handle VP8 allocation.
This includes capturing TemporalLayer instances created by the
encoder.
* ViEEncoder now owns both the bitrate allocator and the temporal layer
factories for VP8. This allows allocation to happen fully outside of
the encoder implementation.
This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.
BUG=webrtc:6301
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2510583002 .
Cr-Commit-Position: refs/heads/master@{#15105}
This approach extends the H.264 specific information with
a packetization mode enum.
Status: Parameter is in code. No way to set it yet.
Rebase of CL 2009213002
BUG=600254
Review-Url: https://codereview.webrtc.org/2337453002
Cr-Commit-Position: refs/heads/master@{#15032}
Reason for revert:
Seems to be causing flakiness in perf test:
FullStackTest.ScreenshareSlidesVP8_2TL_LossyNet
Original issue's description:
> Reland of Issue 2434073003: Extract bitrate allocation ...
>
> This is a reland of https://codereview.webrtc.org/2434073003/ including
> some fixes for failing test cases.
>
> Original description:
>
> Extract bitrate allocation of spatial/temporal layers out of codec impl.
>
> This CL makes a number of intervowen changes:
>
> * Add BitrateAllocation struct, that contains a codec independent view
> of how the target bitrate is distributed over spatial and temporal
> layers.
>
> * Adds the BitrateAllocator interface, which takes a bitrate and frame
> rate and produces a BitrateAllocation.
>
> * A default (non layered) implementation is added, and
> SimulcastRateAllocator is extended to fully handle VP8 allocation.
> This includes capturing TemporalLayer instances created by the
> encoder.
>
> * ViEEncoder now owns both the bitrate allocator and the temporal layer
> factories for VP8. This allows allocation to happen fully outside of
> the encoder implementation.
>
> This refactoring will make it possible for ViEEncoder to signal the
> full picture of target bitrates to the RTCP module.
>
> BUG=webrtc:6301
>
> Committed: https://crrev.com/647bf43dcb2fd16fccf276bd94dc4400728bb405
> Cr-Commit-Position: refs/heads/master@{#15023}
TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2491393002
Cr-Commit-Position: refs/heads/master@{#15026}
This is a reland of https://codereview.webrtc.org/2434073003/ including
some fixes for failing test cases.
Original description:
Extract bitrate allocation of spatial/temporal layers out of codec impl.
This CL makes a number of intervowen changes:
* Add BitrateAllocation struct, that contains a codec independent view
of how the target bitrate is distributed over spatial and temporal
layers.
* Adds the BitrateAllocator interface, which takes a bitrate and frame
rate and produces a BitrateAllocation.
* A default (non layered) implementation is added, and
SimulcastRateAllocator is extended to fully handle VP8 allocation.
This includes capturing TemporalLayer instances created by the
encoder.
* ViEEncoder now owns both the bitrate allocator and the temporal layer
factories for VP8. This allows allocation to happen fully outside of
the encoder implementation.
This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2488833004
Cr-Commit-Position: refs/heads/master@{#15023}
Reason for revert:
Breaks perf tests.
Original issue's description:
> Extract bitrate allocation of spatial/temporal layers out of codec impl.
>
> This CL makes a number of intervowen changes:
>
> * Add BitrateAllocation struct, that contains a codec independent view
> of how the target bitrate is distributed over spatial and temporal
> layers.
>
> * Adds the BitrateAllocator interface, which takes a bitrate and frame
> rate and produces a BitrateAllocation.
>
> * A default (non layered) implementation is added, and
> SimulcastRateAllocator is extended to fully handle VP8 allocation.
> This includes capturing TemporalLayer instances created by the
> encoder.
>
> * ViEEncoder now owns both the bitrate allocator and the temporal layer
> factories for VP8. This allows allocation to happen fully outside of
> the encoder implementation.
>
> This refactoring will make it possible for ViEEncoder to signal the
> full picture of target bitrates to the RTCP module.
>
> BUG=webrtc:6301
>
> Committed: https://crrev.com/8f46c679d24a05b3f08e02c6d91ec9637f34e24f
> Cr-Commit-Position: refs/heads/master@{#14998}
TBR=stefan@webrtc.org,perkj@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:6301
Review-Url: https://codereview.webrtc.org/2489843002
Cr-Commit-Position: refs/heads/master@{#15001}
This CL makes a number of intervowen changes:
* Add BitrateAllocation struct, that contains a codec independent view
of how the target bitrate is distributed over spatial and temporal
layers.
* Adds the BitrateAllocator interface, which takes a bitrate and frame
rate and produces a BitrateAllocation.
* A default (non layered) implementation is added, and
SimulcastRateAllocator is extended to fully handle VP8 allocation.
This includes capturing TemporalLayer instances created by the
encoder.
* ViEEncoder now owns both the bitrate allocator and the temporal layer
factories for VP8. This allows allocation to happen fully outside of
the encoder implementation.
This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2434073003
Cr-Commit-Position: refs/heads/master@{#14998}
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}
I'm also removing media_optimization_unittest.cc, since it only tested the
suspension logic and nothing else.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/2119503002 .
Cr-Commit-Position: refs/heads/master@{#13355}
This cl split the class MediaOptimization into two parts. One that deals with frame dropping and stats and one new class called ProtectionBitrateCalculator that deals with calculating the needed FEC parameters and how much of the estimated network bitrate that can be used by an encoder
Note that the logic of how FEC and the needed bitrates is not changed.
BUG=webrtc:5687
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1972083002 .
Cr-Commit-Position: refs/heads/master@{#13018}
SSRC knowledge is contained withing VideoSendStream. That also means that debug recording is moved to VideoSendStream.
I think that make sence since that allows debug recording with external encoder implementations one day.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1936503002
Cr-Commit-Position: refs/heads/master@{#12632}
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}
Also refactor GenericEncoder to use these file writers, and remove use
of preprocessor to enable file writing.
BUG=
Review URL: https://codereview.webrtc.org/1853813002
Cr-Commit-Position: refs/heads/master@{#12372}
Testing the nack module by implementing it into the current jitter buffer
under the experiment WebRTC-NewVideoJitterBuffer.
BUG=webrtc:5514
Review URL: https://codereview.webrtc.org/1778503002
Cr-Commit-Position: refs/heads/master@{#11969}
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}
