Fix issue where RTCP XR target bitrate could be incorrect when temporal layers are configured but it is not supported by encoder.
Bug: webrtc:10475
Change-Id: Ib525eb5f0ad8392e88d2579930ac8e459a1d194b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128778
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27268}
This CL paves the way to making FrameBufferController injectable.
LibvpxVp8Encoder can manage multiple streams. Prior to this CL,
each stream had its own frame buffer controller, all of them held
in a vector by LibvpxVp8Encoder. This complicated the code and
produced some code duplication (cf. SetupTemporalLayers).
This CL:
1. Replaces CreateVp8TemporalLayers() by a factory. (Later CLs
will make this factory injectable.)
2. Makes LibvpxVp8Encoder use a single controller. This single
controller will, in the case of multiple streams, delegate
its work to multiple controllers, but that fact is not visible
to LibvpxVp8Encoder.
This CL also squashes CL #126046 (Send notifications of RTT and
PLR changes to Vp8FrameBufferController) into it.
Bug: webrtc:10382
Change-Id: Id9b55734bebb457acc276f34a7a9e52cc19c8eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126483
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27206}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: Iedb3be944828a1caba55bbbd4dc0b56c55bbb7d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127624
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27123}
A recent refactoring introduced a subtle difference in how encoded
frame timing is measured. See
https://webrtc-review.googlesource.com/c/src/+/124122/14/video/video_stream_encoder.cc#b1278
After that change, the encode was considered done after encoding was
done. The old behavior included the time needed to call stats and
the video sink, which might include video quality tests related tasks
and RTP packetization.
In order to preserve the old behavior I'm moving timestamping to after
packetization again.
Note that the timing frame info still has a separate timestamp that
does explicitly measure encode time. This is used by the experimental
new overuse detector, so the effect of this change will be transient
anyhow.
Bug: chromium:941457, webrtc:10164
Change-Id: Ia990a1ceaeaf2c45d5df2a32d4f017cdb08e3c55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127569
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27108}
In Chrome we sometimes get frames with native handlers all the time.
To enable variable frame rate we need to trust at least empty updates.
Also, trust empty update rect when scaling.
Bug: webrtc:10310
Change-Id: I6087b66d5e6138290a7c143f85ba9bc427ae40a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126223
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27021}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
This CL plumbs an additional signal from VideoSendStream down to
VideoStreamEncoder, namely the amount of headroom that's left between
the encoder max bitrate and the current bitrate allocation for the
media track.
This will be used in follow-up CLs to tune encoder rate adjustment
and some codec specific paramaters a bit differently, based on the
knowledge if we are network constrained or not.
Bug: webrtc:10155
Change-Id: Ic6ccc79be5c6845468bab65b4ca9918b56923fa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125981
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27008}
This CL takes a few parts of VCMEncodedFrameCallback and
VCMGenericEncoder and folds some aspect directly into
VideoStreamEncoder. Parts related to timing frames are extracted
into a new class FrameEncodeTimer that explicitly handles that.
Bug: webrtc:10164
Change-Id: I9b26f734473b659e4093c84c09fb0ed441290e40
Reviewed-on: https://webrtc-review.googlesource.com/c/124122
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26862}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/123920
Patch set 1 is identical to the previous CL, additional patch sets fix
the bug that was introduced and adds test coverage.
Since this "data base" only holds a single encoder instance it just
serves to confuse object ownership. Removing it and giving ownership
of generic encoder instance to VideoStreamEncoder.
This CL also removes VideoSender interface from video_coding_impl.h,
which is mostly a leftover from
https://webrtc-review.googlesource.com/c/src/+/123540
Bug: webrtc:10164
Change-Id: Ieaf23457d69af0d6356b70461112892b14760b19
Reviewed-on: https://webrtc-review.googlesource.com/c/124488
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26857}
This reverts commit 715c4765b1ac20017e6e3b8b925d02536c6610c3.
Reason for revert: Breaks WebRTC roll to Chromium.
https://chromium-review.googlesource.com/c/chromium/src/+/1484629
# Fatal error in: ../../third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc, line 796
# last system error: 0
# Check failed: diff_ms >= static_cast<int64_t>(0) (-307 vs. 0)
#
Original change's description:
> Remove VCMEncoderDataBase and put remaining code into VideoStreamEncoder
>
> Since this "data base" only holds a single encoder instance it just
> serves to confuse object ownership. Removing it and giving ownership
> of generic encoder instance to VideoStreamEncoder.
>
> This CL also removes VideoSender interface from video_coding_impl.h,
> which is mostly a leftover from
> https://webrtc-review.googlesource.com/c/src/+/123540
>
> Bug: webrtc:10164
> Change-Id: I9b7fec940dbcbccf3aa1278c2555da3bd5169ae1
> Reviewed-on: https://webrtc-review.googlesource.com/c/123920
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26835}
TBR=brandtr@webrtc.org,nisse@webrtc.org,sprang@webrtc.org
Change-Id: I5432878c4c2e497cd848c4ce1b190e0307df03ca
Bug: webrtc:10164
Reviewed-on: https://webrtc-review.googlesource.com/c/124402
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26841}
Since this "data base" only holds a single encoder instance it just
serves to confuse object ownership. Removing it and giving ownership
of generic encoder instance to VideoStreamEncoder.
This CL also removes VideoSender interface from video_coding_impl.h,
which is mostly a leftover from
https://webrtc-review.googlesource.com/c/src/+/123540
Bug: webrtc:10164
Change-Id: I9b7fec940dbcbccf3aa1278c2555da3bd5169ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/123920
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26835}
This CL moves the functionality in VideoSender into VideoStreamEncoder
and simplifies the code where possible, given what we know of the
encoder state and that we now run on the encoder queue.
The intent here is to make it easier to remove the next parts, the
encoder database and generic encoder wrapper.
Bug: webrtc:10164
Change-Id: I8c108ccbe5db97cd9fd1e84228134709af845ea3
Reviewed-on: https://webrtc-review.googlesource.com/c/123540
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26813}
Reland with fixes for failing chromium tests.
Propagate VideoFrame::UpdateRect to encoder
Accumulate it in all places where frames can be dropped before they reach the encoder.
Reset UpdateRect in VideoBroadcaster if frame the previous frame is dropped.
No accumulation is done here since it's supposed to be a brief occasion then configuration have changed.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/123102
Bug: webrtc:10310
Change-Id: I18be73f47f227d6392bf9cb220b549ced225714f
Reviewed-on: https://webrtc-review.googlesource.com/c/123230
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26738}
Accumulate it in all places where frames can be dropped before they reach
the encoder.
Reset UpdateRect in VideoBroadcaster if frame the previous frame is dropped.
No accumulation is done here since it's supposed to be a brief occusion then
configuration have changed.
Bug: webrtc:10310
Change-Id: I2813ecd009eb730bd99ffa0a02f979091b56bf80
Reviewed-on: https://webrtc-review.googlesource.com/c/123102
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26711}
The overshoot detector uses a simple pacer model to determine an
estimate of how much the encoder is overusing the target bitrate.
This utilization factor can then be adjuster for when configuring the
actual target bitrate.
Spatial layers (simulcast streams) are adjusted separately.
Temporal layers are measured separately, but are combined into a single
utilization factor per spatial layer.
Bug: webrtc:10155
Change-Id: I8ea58dc6c4871e880553d7c22202f11cb2feb216
Reviewed-on: https://webrtc-review.googlesource.com/c/114886
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26573}
Currently, CPU overuse settings for HW encoders are sometimes being used
even though the actual encoder is a SW encoder, e.g. in case of SW fallback
when the encoder is initialized. Polling is_hardware_accelerated after the
encoder has been created and initialized will improve choosing the correct
CPU overuse settings.
Bug: webrtc:10065
Change-Id: Ic6bd67630a040b5a121c13fa63dd074006973929
Reviewed-on: https://webrtc-review.googlesource.com/c/116688
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26266}
This reverts commit 6613f8e98ab3654ade7e8f5352d8d6711b157499.
Reason for revert: This change seemed innocent after all, so undoing speculative revert.
Original change's description:
> Revert "Refactor and remove media_optimization::MediaOptimization."
>
> This reverts commit 07276e4f89a93b1479d7aeefa53b4fc32daf001b.
>
> Reason for revert: Speculative revert due to downstream crashes.
>
> Original change's description:
> > Refactor and remove media_optimization::MediaOptimization.
> >
> > This CL removes MediaOptmization and folds some of its functionality
> > into VideoStreamEncoder.
> >
> > The FPS tracking is now handled by a RateStatistics instance. Frame
> > dropping is still handled by FrameDropper. Both of these now live
> > directly in VideoStreamEncoder.
> > There is no intended change in behavior from this CL, but due to a new
> > way of measuring frame rate, some minor perf changes can be expected.
> >
> > A small change in behavior is that OnBitrateUpdated is now called
> > directly rather than on the next frame. Since both encoding frame and
> > setting rate allocations happen on the encoder worker thread, there's
> > really no reason to cache bitrates and wait until the next frame.
> > An edge case though is that if a new bitrate is set before the first
> > frame, we must remember that bitrate and then apply it after the video
> > bitrate allocator has been first created.
> >
> > In addition to existing unit tests, manual tests have been used to
> > confirm that frame dropping works as expected with misbehaving encoders.
> >
> > Bug: webrtc:10164
> > Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
> > Reviewed-on: https://webrtc-review.googlesource.com/c/115620
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26147}
>
> TBR=nisse@webrtc.org,sprang@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10164
> Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c
> Reviewed-on: https://webrtc-review.googlesource.com/c/116780
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26191}
TBR=nisse@webrtc.org,sprang@webrtc.org
Change-Id: Ieda1fad301de002460bb0bf5a75267ea065176a8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10164
Reviewed-on: https://webrtc-review.googlesource.com/c/116960
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26213}
This reverts commit 07276e4f89a93b1479d7aeefa53b4fc32daf001b.
Reason for revert: Speculative revert due to downstream crashes.
Original change's description:
> Refactor and remove media_optimization::MediaOptimization.
>
> This CL removes MediaOptmization and folds some of its functionality
> into VideoStreamEncoder.
>
> The FPS tracking is now handled by a RateStatistics instance. Frame
> dropping is still handled by FrameDropper. Both of these now live
> directly in VideoStreamEncoder.
> There is no intended change in behavior from this CL, but due to a new
> way of measuring frame rate, some minor perf changes can be expected.
>
> A small change in behavior is that OnBitrateUpdated is now called
> directly rather than on the next frame. Since both encoding frame and
> setting rate allocations happen on the encoder worker thread, there's
> really no reason to cache bitrates and wait until the next frame.
> An edge case though is that if a new bitrate is set before the first
> frame, we must remember that bitrate and then apply it after the video
> bitrate allocator has been first created.
>
> In addition to existing unit tests, manual tests have been used to
> confirm that frame dropping works as expected with misbehaving encoders.
>
> Bug: webrtc:10164
> Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
> Reviewed-on: https://webrtc-review.googlesource.com/c/115620
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26147}
TBR=nisse@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10164
Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c
Reviewed-on: https://webrtc-review.googlesource.com/c/116780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26191}
This CL removes MediaOptmization and folds some of its functionality
into VideoStreamEncoder.
The FPS tracking is now handled by a RateStatistics instance. Frame
dropping is still handled by FrameDropper. Both of these now live
directly in VideoStreamEncoder.
There is no intended change in behavior from this CL, but due to a new
way of measuring frame rate, some minor perf changes can be expected.
A small change in behavior is that OnBitrateUpdated is now called
directly rather than on the next frame. Since both encoding frame and
setting rate allocations happen on the encoder worker thread, there's
really no reason to cache bitrates and wait until the next frame.
An edge case though is that if a new bitrate is set before the first
frame, we must remember that bitrate and then apply it after the video
bitrate allocator has been first created.
In addition to existing unit tests, manual tests have been used to
confirm that frame dropping works as expected with misbehaving encoders.
Bug: webrtc:10164
Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
Reviewed-on: https://webrtc-review.googlesource.com/c/115620
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26147}
* Stop encoding spatial layers S(n >= N) if application deactivates
spatial layer N by setting RTCRtpEncodingParameters.active = false.
* Move calculation of padding bitrate to SvcRateAllocator class.
* Pad up to minimum required bitrate of base layer if ALR probing is
enabled.
Bug: webrtc:9350
Change-Id: I398284c943d43348def535c83263fc234c9767fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25945}
The current code assumes that layers are ordered from the smallest
to the largest.
If that assumption is broken and the last layer is smaller than the
others, all layers that are bigger will be scaled up.
Bug: webrtc:10069
Change-Id: Iff87ddba741d5dfe3d0cc25a8f75d898a417eec7
Reviewed-on: https://webrtc-review.googlesource.com/c/112460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25878}
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
This deprecates the following methods in VideoEncoder:
virtual ScalingSettings GetScalingSettings() const;
virtual bool SupportsNativeHandle() const;
virtual const char* ImplementationName() const;
Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.
Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().
This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.
Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
