Reason for revert:
Breaks Chrome FYI using H264.
Need to investigate.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win7%20Tester/builds/4170
Original issue's description:
> Remove ViEEncoder::SetNetworkStatus
>
> This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
>
> BUG=webrtc:5687
> NOTRY=True
>
> Committed: https://crrev.com/50b5c3be844ef571a28b2681c549443a26735d72
> Cr-Commit-Position: refs/heads/master@{#12699}
TBR=stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1978783002
Cr-Commit-Position: refs/heads/master@{#12715}
This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
BUG=webrtc:5687
NOTRY=True
Review-Url: https://codereview.webrtc.org/1932683002
Cr-Commit-Position: refs/heads/master@{#12699}
This reverts commit e30c27205148b34ba421184efe65f6a0780b436d (https://codereview.webrtc.org/1958053002/)
Original reverted cl is in patch set #1.
Changes in following patch sets.
The cl now also make sure SendPacer starts with the configured bitrate provided in a call to CongestionController::SetBweBitrates)()
It turns out that the failing tests in 609816 is due to a bug in the current code that runs the proper at 300kbit regardless of configured start bitrate.
Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
BUG=chromium:609816, webrtc:5687
TBR=mflodman@webrtc.org
NOTRY=True // Due to bug in android_x86 cq builder....
Review-Url: https://codereview.webrtc.org/1958113003
Cr-Commit-Position: refs/heads/master@{#12688}
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}
This reverts commit 825eb58d59940a4c3c9837595c4b3b07059c93ca.
This Relands the cl reviewed in https://codereview.webrtc.org/1917793002/
patchset #1 is a pure reland.
patchset #2 fix an overflow in BitrateProber that caused WebRtcVideoChannel2BaseTest.TwoStreamsSendAndReceive to fail.
Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
BUG=webrtc:5687
Review URL: https://codereview.webrtc.org/1947873002 .
Cr-Commit-Position: refs/heads/master@{#12630}
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator)
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1917793002
Cr-Commit-Position: refs/heads/master@{#12620}
- "WebRTC.Video.SendDelayInMs"
Change so that PacketOption packet id is always set in RtpSender (if having a TransportSequenceNumberAllocator).
Add SendDelayStats class for computing delays.
Add SendPacketObserver to RtpRtcp config and register SendDelayStats as observer.
Wire up OnSentPacket to SendDelayStats.
BUG=webrtc:5215
Review-Url: https://codereview.webrtc.org/1478253002
Cr-Commit-Position: refs/heads/master@{#12600}
And move encoder name cb to VCMSendStatisticsCallback.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1900193004
Cr-Commit-Position: refs/heads/master@{#12596}
ViEEncoder doesn't need a full VideoCodingModule since it only uses the
sender side either way.
BUG=webrtc:3608,webrtc:5687
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1904983002 .
Cr-Commit-Position: refs/heads/master@{#12456}
Reason for revert:
RTCVideoEncoder has been updated to not make assumptions on calling threads/post back to a worker thread. This should now be landable again.
Original issue's description:
> Revert of Initialize/configure video encoders asychronously. (patchset #4 id:60001 of https://codereview.webrtc.org/1757313002/ )
>
> Reason for revert:
> Breaks RTCVideoEncoder which has incorrect assumptions on where InitEncode etc. is called from. Temporarily reverting until RTCVideoEncoder has been updated.
>
> Original issue's description:
> > Initialize/configure video encoders asychronously.
> >
> > Greatly speeds up setRemoteDescription() by moving encoder initialization
> > off the main worker thread, which is free to move onto gathering ICE
> > candidates and other tasks while InitEncode() is performed. It also
> > un-blocks PeerConnection GetStats() which is no longer blocked on
> > encoder initialization.
> >
> > BUG=webrtc:5410
> > R=stefan@webrtc.org
> >
> > Committed: fb647a67be
>
> R=stefan@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:595274, chromium:595308, webrtc:5410
>
> Committed: https://crrev.com/81cbd924447d507559dbd6e6d1f9fe439fcf2716
> Cr-Commit-Position: refs/heads/master@{#12086}
TBR=stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:595274, chromium:595308, webrtc:5410
Review URL: https://codereview.webrtc.org/1896413002
Cr-Commit-Position: refs/heads/master@{#12446}
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}
Reason for revert:
Breaks RTCVideoEncoder which has incorrect assumptions on where InitEncode etc. is called from. Temporarily reverting until RTCVideoEncoder has been updated.
Original issue's description:
> Initialize/configure video encoders asychronously.
>
> Greatly speeds up setRemoteDescription() by moving encoder initialization
> off the main worker thread, which is free to move onto gathering ICE
> candidates and other tasks while InitEncode() is performed. It also
> un-blocks PeerConnection GetStats() which is no longer blocked on
> encoder initialization.
>
> BUG=webrtc:5410
> R=stefan@webrtc.org
>
> Committed: fb647a67beR=stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:595274, chromium:595308, webrtc:5410
Review URL: https://codereview.webrtc.org/1821983002 .
Cr-Commit-Position: refs/heads/master@{#12086}
It was possible that even after a VideoSendStream was destroyed,
it remained registered as a BitrateAllocator observer, causing a
segfault later.
Review URL: https://codereview.webrtc.org/1815733002
Cr-Commit-Position: refs/heads/master@{#12067}
Greatly speeds up setRemoteDescription() by moving encoder initialization
off the main worker thread, which is free to move onto gathering ICE
candidates and other tasks while InitEncode() is performed. It also
un-blocks PeerConnection GetStats() which is no longer blocked on
encoder initialization.
BUG=webrtc:5410
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1757313002 .
Cr-Commit-Position: refs/heads/master@{#11983}
This CL will be followed up with a CL adding AudioSendStream to
BitrateAllocator, so this is a small CL to have the video connection to
BitrateAllocator "at the same level" as for audio.
BUG=webrtc:5079
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1785283002 .
Cr-Commit-Position: refs/heads/master@{#11955}
Makes VideoCaptureInput easier to test and enables running more things
outside VideoCaptureInput on the encoder thread in the future
(initializing encoders and reconfiguring them, for instance).
BUG=webrtc:5410, webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1763693002 .
Cr-Commit-Position: refs/heads/master@{#11860}
Intended to make SetEncoder callable from another thread so that
ReconfigureVideoEncoder can post SetEncoder over and return earlier to
prevent blocking the calling thread.
BUG=webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1751363003 .
Cr-Commit-Position: refs/heads/master@{#11856}
Also moves and simplifies SetSendCodec from VideoSendStream to mostly
inside ViEEncoder. This is necessary for making
ReconfigureVideoEncoder asynchronous as we don't post any result back.
BUG=webrtc:5494
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1754283002 .
Cr-Commit-Position: refs/heads/master@{#11847}
Removes StartSend, StopSend and SetSendCodec from ViEChannel and into
VideoSendStream which uses the payload router to configure them
directly.
BUG=webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1758603003 .
Cr-Commit-Position: refs/heads/master@{#11845}
This cl copies the value of cricket::VideoCapturer::IsScreencast into
a flag in VideoOptions. It is passed on via the chain
VideortpSender::SetVideoSend
WebRtcVideoChannel2::SetVideoSend
WebRtcVideoChannel2::SetOptions
WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions
Where it's used, in
WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame, we can look it up
in parameters_, instead of calling capturer_->IsScreencast().
Doesn't touch screencast logic related to cpu adaptation, since that
code is in flux in a different cl.
Also drop the is_screencast flag from the Dimensions struct, and drop separate options argument from ConfigureVideoEncoderSettings and SetCodecAndOptions, instead always using the options recorded in VideoSendStreamParameters::options.
In the tests, changed FakeVideoCapturer::is_screencast to be a construction time flag. Generally, unittests of screencast have to both use a capturer configured for screencast, and set the screencast flag using SetSendParameters. Since the automatic connection via VideoSource and VideoRtpSender isn't involved in the unit tests.
Note that using SetSendParameters to set the screencast flag doesn't make sense, since it's not per-stream. SetVideoSend would be more appropriate. That should be fixed if/when we drop VideoOptions from SetSendParameters.
BUG=webrtc:5426
R=pbos@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1711763003 .
Cr-Commit-Position: refs/heads/master@{#11837}
Removes per-extension functions in ViEChannel/ViEReceiver and instead
register extensions directly on the RTP module by mapping extension
string to RTP-header-extension type.
BUG=webrtc:5494
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1740133002 .
Cr-Commit-Position: refs/heads/master@{#11786}
Puts thresholds in a range that works well on Nexus 5X (doesn't
seem to trigger overuse), while not disabling them for systems that have
a really-really hard time (>200% overuse).
BUG=webrtc:5577
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1730103003 .
Cr-Commit-Position: refs/heads/master@{#11744}
This allows other projects to more easily depend on this.
The plan is to move remote_bitrate_estimator and bitrate_controller into this module and reduce the exposed interface to only a simplified version of congestion_controller.h.
No functional changes in this CL.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1718473002 .
Cr-Commit-Position: refs/heads/master@{#11718}
Moves RtpRtcp module pointers into VideoSendStream and uses them for
simple calls that were only forwarded by ViEChannel.
BUG=webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1693553002 .
Cr-Commit-Position: refs/heads/master@{#11709}
Since SSRCs can no longer change on the fly, SSRC code can be made a lot
simpler (and faster). Resulting code has less and shorter locking.
BUG=webrtc:5494
R=danilchap@webrtc.org
Review URL: https://codereview.webrtc.org/1713683003 .
Cr-Commit-Position: refs/heads/master@{#11691}
Also move some stats reporting from vie_channel to send stats proxy
BUG=
Review URL: https://codereview.webrtc.org/1669623004
Cr-Commit-Position: refs/heads/master@{#11688}
EncoderStateFeedback is now only connected to one encoder, so remove map
and other complexity to deliver feedback more directly.
BUG=webrtc:5494
R=danilchap@webrtc.org
Review URL: https://codereview.webrtc.org/1706803002 .
Cr-Commit-Position: refs/heads/master@{#11687}
Reason for revert:
Disabling tests on memcheck that time out due to using real VP8 encoders.
Original issue's description:
> Revert of Don't send FEC for H.264 with NACK enabled. (patchset #5 id:80001 of https://codereview.webrtc.org/1687303002/ )
>
> Reason for revert:
> Broke the VerifyHistogramStatsWithRed test on the Windows DrMemory Full bot and Linux Memcheck bot. Please fix the test and reland.
>
> Original issue's description:
> > Don't send FEC for H.264 with NACK enabled.
> >
> > The H.264 does not contain picture IDs and are not sufficient to
> > determine that a packet may be skipped. This causes retransmission
> > requests for FEC that are currently dropped by the sender (since they
> > should be redundant).
> >
> > The receiver is then unable to continue without having the packet gap
> > filled (unlike VP8/VP9 which moves on since it has a consecutive stream
> > of picture IDs).
> >
> > Even if FEC retransmission did work it's a huge waste of bandwidth,
> > since it just adds additional overhead that has to be unconditionally
> > transmitted. This bandwidth is better used to send higher-quality
> > frames.
> >
> > BUG=webrtc:5264
> > R=stefan@webrtc.org
> >
> > Committed: https://crrev.com/25558ad819b4df41ba51537e26a77480ace1e525
> > Cr-Commit-Position: refs/heads/master@{#11601}
>
> TBR=stefan@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5264
>
> Committed: https://crrev.com/29ffdc1a15e31bd81e806ff135c2100d811714f0
> Cr-Commit-Position: refs/heads/master@{#11607}
TBR=stefan@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5264
Review URL: https://codereview.webrtc.org/1697093002 .
Cr-Commit-Position: refs/heads/master@{#11621}
Reason for revert:
Broke the VerifyHistogramStatsWithRed test on the Windows DrMemory Full bot and Linux Memcheck bot. Please fix the test and reland.
Original issue's description:
> Don't send FEC for H.264 with NACK enabled.
>
> The H.264 does not contain picture IDs and are not sufficient to
> determine that a packet may be skipped. This causes retransmission
> requests for FEC that are currently dropped by the sender (since they
> should be redundant).
>
> The receiver is then unable to continue without having the packet gap
> filled (unlike VP8/VP9 which moves on since it has a consecutive stream
> of picture IDs).
>
> Even if FEC retransmission did work it's a huge waste of bandwidth,
> since it just adds additional overhead that has to be unconditionally
> transmitted. This bandwidth is better used to send higher-quality
> frames.
>
> BUG=webrtc:5264
> R=stefan@webrtc.org
>
> Committed: https://crrev.com/25558ad819b4df41ba51537e26a77480ace1e525
> Cr-Commit-Position: refs/heads/master@{#11601}
TBR=stefan@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5264
Review URL: https://codereview.webrtc.org/1692783005
Cr-Commit-Position: refs/heads/master@{#11607}
Removes protection-callback removal and makes ViEChannel outlive
ViEEncoder to not have races between BitrateAllocator and
VideoSendStream destruction.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1696693002 .
Cr-Commit-Position: refs/heads/master@{#11604}
The H.264 does not contain picture IDs and are not sufficient to
determine that a packet may be skipped. This causes retransmission
requests for FEC that are currently dropped by the sender (since they
should be redundant).
The receiver is then unable to continue without having the packet gap
filled (unlike VP8/VP9 which moves on since it has a consecutive stream
of picture IDs).
Even if FEC retransmission did work it's a huge waste of bandwidth,
since it just adds additional overhead that has to be unconditionally
transmitted. This bandwidth is better used to send higher-quality
frames.
BUG=webrtc:5264
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1687303002 .
Cr-Commit-Position: refs/heads/master@{#11601}
Remove hops into ViEChannel for calls directly into RtpRtcp and
ViEReceiver from VideoReceiveStream.
Some calls are more complex and will be removed later.
BUG=webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1671893002 .
Cr-Commit-Position: refs/heads/master@{#11526}
