Commit Graph

241 Commits

Author SHA1 Message Date
b4e96d48a2 VideoStreamEncoder: Introduce frame cadence adapter.
This change introduces a new FrameCadenceAdapter class which takes the
role of being a VideoFrameSinkInterface<> instead of VideoStreamEncoder.
The FrameCadenceAdapter will see its functionality grow in future CLs
and eventually enable screenshare capture sources to have zero hertz as
the minimum capture frequency.

This CL moves logic related to UMA collection and constraints into the
adapter.

The adapter has two major modes. Future functionality is planned to be
added under the WebRTC-ZeroHertzScreenshare field trial. Unit tests are
added that verify passthrough operation when WebRTC-ZeroHertzScreenshare
isn't specified or disabled.

Just specifying the WebRTC-ZeroHertzScreenshare field trial isn't
enough to activate the feature, but the caller has to additionally
configure screen content type, minimum FPS 0, and maximum FPS > 0 for
the new mode.

go/rtc-0hz-present

Bug: chromium:1255737
Change-Id: I1799110ed40843152786ad80df10acfb83a608b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236682
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35315}
2021-11-05 12:37:45 +00:00
97597c0f51 Remove usage of INFO alias for LS_INFO in log messages
Bug: webrtc:13362
Change-Id: Ifda893861a036a85c045cd366f9eab33c62ebde0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237221
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35310}
2021-11-04 13:46:17 +00:00
2b10c479ce VideoStreamEncoder: clean up threading constraints.
The sequences of threads entering the VideoStreamEncoder has been
unclear. Fix this by renaming the uninformational |main_queue_| to
|worker_queue_|, and introduce a new |network_queue_| which is set
on construction.

Bug: chromium:1255737
Change-Id: Ic4d3a5b8188b8cc98e60b72aee2c09c9afbc7356
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236523
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35283}
2021-10-29 12:05:11 +00:00
606d3cb1cf VideoStreamEncoderTest: Use DataRate for some constants.
Use config from FakeEncoder in some tests.

Bug: none
Change-Id: I1d7e01f604f8aabb5d6815bb519ef2532d024d76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233243
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35138}
2021-10-04 10:10:40 +00:00
f270770679 video: Implement bandwidth based scaler
The |slice_qp_detla| reported by the hardware is not credible, which
causing the quality scaler cannot work properly,the resolution cannot
be adjusted correctly.

To fix this issue, this CL implements a bandwidth scaler which is used
for adjust resolution, this scaler will be used when QP based quality
scaler is not working due to untrusted QP reported by HW AVC encoder.

Bug: webrtc:12942
Change-Id: I2fc5f07a5400ec7e5ead2c2c502faee84d7f2a76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35120}
2021-09-29 10:39:27 +00:00
4d4f62f6e7 VideoSendStreamTest: Add tests for encoder reconfiguration.
Bug: none
Change-Id: I1d976eb77357c7050ed6ca7d0eee9153f9ef0251
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231000
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34978}
2021-09-13 13:14:22 +00:00
06defc4320 QualityRampupExperiment: SetMaxBitrate may not be set correctly.
Call SetMaxBitrate when encoder is configured instead of in OnMaybeEncodeFrame (which is called after the initial frame dropping ->
max bitrate is not set for dropped frames).

Added support for single active stream configuration.

Bug: none
Change-Id: I33ff96e7feed70b9ea3c9b3da89f117859108347
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231681
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34973}
2021-09-11 10:28:43 +00:00
b54cfdebfe Add optional is_qp_trusted property for EncoderInfo.
Some hardware H.264 encoders does not place average QP delta in
slice_qp_delta field. Adding an optional flag in EncoderInfo to notify
quality scaler about this.

Bug: webrtc:12942
Change-Id: I3ee29c5ae9bd7bb34d26eba7e6bede3798ca44b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226921
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34627}
2021-08-02 13:49:21 +00:00
ab30d72b72 Use backticks not vertical bars to denote variables in comments for /video
Bug: webrtc:12338
Change-Id: I47958800407482894ff6f17c1887dce907fdf35a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227030
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34585}
2021-07-28 13:22:27 +00:00
8b6929081e Fix VideoStreamEncoder QP tests to not use SetHasInternalSource
The has_internal_source feature is deprecated, and unrelated to the
tests of QP parsing.

Bug: webtc:12875
Change-Id: Ib43063ebf49e6e0bd7a5328a04ba2816f3a7ecb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222400
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34280}
2021-06-14 14:07:46 +00:00
f5f7e8e806 Ensure that fps adaptation count can go back to zero when framerate is unrestricted.
Bug: webrtc:12867
Change-Id: I1c11d1a1154ea3d802cdc01e260f72a7e9d17e99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221373
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34265}
2021-06-10 16:00:39 +00:00
7444b19833 Add integration test for active stream toggling.
Bug: webrtc:12778
Change-Id: I0441d05daef0b2003e6a5710c7a2b30978ffb6ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220930
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34201}
2021-06-02 13:51:35 +00:00
c1d589146b Replace new rtc::RefCountedObject with rtc::make_ref_counted in a few files
Bug: webrtc:12701
Change-Id: Ie50225374f811424faf20caf4cf454b2fd1c4dc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215930
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33818}
2021-04-23 12:04:39 +00:00
9071957da3 Remove unused members in tests.
VideoStreamEncoderTest: Remove unneeded set_timestamp_rtp in CreateFrame methods (the timestamp is set based on ntp_time_ms in VideoStreamEncoder::OnFrame).

Bug: none
Change-Id: I6b5531a9ac21cde5dac54df6de9b9d43261e90c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214488
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33683}
2021-04-12 07:21:03 +00:00
73cf80a932 Fixes incorrect feedback to EncoderBitrateAdjuster [perf note]
At the point where an EncodedImage is reported to the
EncoderBitrateAdjuster (in order to estimate utilization), the image
data has been cleared so the size is 0 - meaning the esimtated
utilization is 0 so pushback is in effect only applied at the
beginning before an estimate is available.

This CL fixes that by explicitly using spatial/temporal id and size in
bytes, rather than passing along the EncodedImage proxy.

It is unclear when this broke, but the regression seems rather old.

This CL will affect the encoded bitrate (and thus indirectly BWE
ramp-up rate), but should avoid exessive delay at low bitrates.
Perf bots will likely trigger alerts, this is expected.

In case there are undesired side-effects, we can entirely disable the
adjuster using existing field-trials.

Bug: webrtc:12606
Change-Id: I936c2045f554696d8b4bb518eee6871ffc12c47d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212900
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33550}
2021-03-24 12:08:23 +00:00
56db9ff1e1 VideoStreamEncoder: Don't map kNative video frame buffers.
Follow-up CL to VP8 and VP9 encoders taking care of mapping.
Context again:
  This CL is part of Optimized Scaling efforts. In Chromium, the native
frame buffer is getting an optimized CropAndScale() implementation. To
support HW accelerated scaling, returning pre-scaled images and skipping
unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale
instead of libyuv::XXXXScale and 2) only map buffers it actually intends
to encode.

In this CL, VideoStreamEncoder no longer calls GetMappedFrameBuffer() on
behalf of the encoders, since the encoders are now able to either do the
mapping or performs ToI420() anyway.

- Tests for old VSE behaviors are updated to test the new behavior (i.e.
  that native frames are pretty much always forwarded).
- The "having to call ToI420() twice" workaround to Android bug
  https://crbug.com/webrtc/12602 is added to H264 and AV1 encoders.

Bug: webrtc:12469
Change-Id: Ibdc2e138d4782a140f433c8330950e61b9829f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211940
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#33548}
2021-03-24 09:43:11 +00:00
d19e3b9676 Reland "Reland "Enable quality scaling when allowed""
This reverts commit 31c5c9da35209fe65ed15cb3a804823cd2789259.

Reason for revert: made QP parser thread-safe https://webrtc.googlesource.com/src/+/0e42cf703bd111fde235d06d08b02d3a7b02c727

Original change's description:
> Revert "Reland "Enable quality scaling when allowed""
>
> This reverts commit 0021fe77937f386e6021a5451e3b0d78d7950815.
>
> Reason for revert: Broken on linux_tsan bot: https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview
>
> Original change's description:
> > Reland "Enable quality scaling when allowed"
> >
> > This reverts commit eb449a979bc561a8b256cca434e582f3889375e2.
> >
> > Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508
> >
> > Original change's description:
> > Before this CL quality scaling was conditioned on scaling settings
> > provided by encoder. That should not be a requirement since encoder
> > may not be aware of quality scaling which is a WebRTC feature. In M90
> > chromium HW encoders do not provide scaling settings (chromium:1179020).
> > The default scaling settings provided by these encoders are not correct
> > (b/181537172).
> >
> > This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> > is set to true in singlecast with normal video feed (not screen sharing)
> > mode. If quality scaling is allowed it is enabled no matter whether
> > scaling settings are present in encoder info or not. Setting from
> > QualityScalingExperiment are used in case if not provided by encoder.
> >
> > Bug: chromium:1179020
> > Bug: webrtc:12511
> > Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33438}
>
> TBR=brandtr@webrtc.org,ilnik@webrtc.org,ssilkin@webrtc.org,rubber-stamper@appspot.gserviceaccount.com
>
> Change-Id: Id7633a1e98f95762e81487887f83a0c35f89195c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1179020
> Bug: webrtc:12511
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211352
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33439}

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I3a31e1c1fdf7da93226f8c1e0a96b43fe0b786df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212026
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33481}
2021-03-16 15:13:52 +00:00
0e42cf703b Reland "Parse encoded frame QP if not provided by encoder"
This reverts commit 727d2afc4330efebc904e0e4f366e885d7b08787.

Reason for revert: Use thread-safe wrapper for H264 parser.

Original change's description:
> Revert "Parse encoded frame QP if not provided by encoder"
>
> This reverts commit 8639673f0c098efc294a7593fa3bd98e28ab7508.
>
> Reason for revert: linux_tsan fails https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview
>
> Original change's description:
> > Parse encoded frame QP if not provided by encoder
> >
> > Bug: webrtc:12542
> > Change-Id: Ic70b46e226f158db7a478a9f20e1f940804febba
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210966
> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33434}
>
> TBR=asapersson@webrtc.org,ssilkin@webrtc.org
>
> Change-Id: Ie251d8f70f8e87fd86b63730aefd2ef3f941e4bb
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12542
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211355
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33441}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:12542
Change-Id: Ib7601fd6f2f26bceddbea2b4ba54d67a281f3a59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211660
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33458}
2021-03-15 10:11:22 +00:00
fd87944042 Removed WebRTC-NetworkCondition-EncoderSwitch field trial.
Bug: webrtc:12474
Change-Id: I50b3219c0dc9d8a63ff097ee6a71c04fe903aea9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211663
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33449}
2021-03-12 16:12:55 +00:00
727d2afc43 Revert "Parse encoded frame QP if not provided by encoder"
This reverts commit 8639673f0c098efc294a7593fa3bd98e28ab7508.

Reason for revert: linux_tsan fails https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview 

Original change's description:
> Parse encoded frame QP if not provided by encoder
>
> Bug: webrtc:12542
> Change-Id: Ic70b46e226f158db7a478a9f20e1f940804febba
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210966
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33434}

TBR=asapersson@webrtc.org,ssilkin@webrtc.org

Change-Id: Ie251d8f70f8e87fd86b63730aefd2ef3f941e4bb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12542
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211355
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33441}
2021-03-11 17:06:06 +00:00
31c5c9da35 Revert "Reland "Enable quality scaling when allowed""
This reverts commit 0021fe77937f386e6021a5451e3b0d78d7950815.

Reason for revert: Broken on linux_tsan bot: https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview

Original change's description:
> Reland "Enable quality scaling when allowed"
>
> This reverts commit eb449a979bc561a8b256cca434e582f3889375e2.
>
> Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508
>
> Original change's description:
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020
> Bug: webrtc:12511
> Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33438}

TBR=brandtr@webrtc.org,ilnik@webrtc.org,ssilkin@webrtc.org,rubber-stamper@appspot.gserviceaccount.com

Change-Id: Id7633a1e98f95762e81487887f83a0c35f89195c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1179020
Bug: webrtc:12511
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211352
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33439}
2021-03-11 15:14:42 +00:00
0021fe7793 Reland "Enable quality scaling when allowed"
This reverts commit eb449a979bc561a8b256cca434e582f3889375e2.

Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508

Original change's description:
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).

This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.

Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33438}
2021-03-11 13:43:11 +00:00
8639673f0c Parse encoded frame QP if not provided by encoder
Bug: webrtc:12542
Change-Id: Ic70b46e226f158db7a478a9f20e1f940804febba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210966
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33434}
2021-03-11 11:48:38 +00:00
eb449a979b Revert "Reland "Enable quality scaling when allowed""
This reverts commit 83be84bb74133343358bba22e4e5106ecc385721.

Reason for revert: Suspect of crbug.com/1185276

Original change's description:
> Reland "Enable quality scaling when allowed"
>
> This reverts commit 609b524dd3ff36719b5c4470b85d37dcdadfb1f8.
>
> Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.
>
> Original change's description:
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020
> Bug: webrtc:12511
> Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33385}

Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I7004014c5936176f8c125aeb55da91ce095b266e
TBR: ssilkin@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209708
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33393}
2021-03-06 09:40:50 +00:00
a86b29be01 Add VP9-specific default resolution bitrate limits
Bug: none
Change-Id: Ifb6f962f04b1f05d20f80a721b1f41904e0a7e99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209702
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33389}
2021-03-05 14:28:14 +00:00
83be84bb74 Reland "Enable quality scaling when allowed"
This reverts commit 609b524dd3ff36719b5c4470b85d37dcdadfb1f8.

Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.

Original change's description:
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).

This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.

Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33385}
2021-03-04 16:01:23 +00:00
609b524dd3 Revert "Enable quality scaling when allowed"
This reverts commit 752cbaba907de077e5f1b24a232e71feb479dccb.

Reason for revert: The test VideoStreamEncoderTest.QualityScalingAllowed_QualityScalingEnabled seems to fail on iOS.

Original change's description:
> Enable quality scaling when allowed
>
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020, webrtc:12511
> Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33373}

Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Icabf2d9a034d359f79491f2c37f1044f17a7445d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209641
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33381}
2021-03-04 10:11:36 +00:00
752cbaba90 Enable quality scaling when allowed
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).

This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.

Bug: chromium:1179020, webrtc:12511
Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33373}
2021-03-03 13:57:22 +00:00
258e9899f4 Use default ResolutionBitrateLimits for simulcast with one active stream if not configured
Bug: none
Change-Id: I049dd0924adc43ce249a8eda63cdcb13da42b030
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208541
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33343}
2021-02-25 12:14:45 +00:00
1124ed1ab2 Communicate encoder resolutions via rtc::VideoSinkWants.
This will allow us to optimize the internal buffers of
webrtc::VideoFrame for the resolution(s) that we actually want to
encode.

Bug: webrtc:12469, chromium:1157072
Change-Id: If378b52b5e35aa9a9800c1f7dfe189437ce43253
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33342}
2021-02-25 11:10:55 +00:00
483b31c231 Reland "Enable Video-QualityScaling experiment by default"
This time exclude iOS from the default behaviour.

Bug: webrtc:12401
Change-Id: Ib1f77123b72c3365591b56455332b3d97b307b26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205006
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33173}
2021-02-05 09:49:13 +00:00
7f354f8606 Use bandwidth allocation in DropDueToSize when incoming resolution increases.
Use bandwidth allocation instead of encoder target bitrate in DropDueToSize when incoming resolution increases to avoid downgrades due to target bitrate being limited by the max bitrate at low resolutions.

Bug: none
Change-Id: Ic41b31c1a86911d4e97b61b0cbc41ce0da739bd4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205622
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33168}
2021-02-04 17:26:21 +00:00
d6604df27f Revert "Enable Video-QualityScaling experiment by default"
This reverts commit 066b5b6ed7069d78e17b8ad6fb8c82546b31acea.

Reason for revert: Regressions on iOS testbots.

Original change's description:
> Enable Video-QualityScaling experiment by default
>
> Bug: webrtc:12401
> Change-Id: Iebf3130e785892bb9fddf1012bc46027a21085a4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204000
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33091}

TBR=ilnik@webrtc.org,asapersson@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:12401
Change-Id: I489b805c7741b63c22c16cfce03347179a3e2602
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205001
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33123}
2021-02-01 13:20:49 +00:00
c91c4233e3 LibvpxVp9Encoder: add option to configure resolution_bitrate_limits.
Bug: none
Change-Id: Icdd7333296d652b1e0c159226df702084303475c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204701
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33121}
2021-02-01 11:48:00 +00:00
066b5b6ed7 Enable Video-QualityScaling experiment by default
Bug: webrtc:12401
Change-Id: Iebf3130e785892bb9fddf1012bc46027a21085a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204000
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33091}
2021-01-28 18:17:03 +00:00
461b1d903f Restart CPU overuse detection when encoder settings has changed.
This could potentially lead to unnecessary restarts since it is also
started after the encoder is created. However, it is needed since the
hardware acceleration support can change even though the encoder has
not been recreated.

Bug: b/145730598
Change-Id: Iad1330e7c7bdf769a68c4ecf7abb6abbf3a4fe71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203140
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33060}
2021-01-22 17:10:12 +00:00
a7e34d33fe Add resolution_bitrate_limits to EncoderInfo field trial.
Added class EncoderInfoSettings for parsing settings.
Added use of class to SimulcastEncoderAdapter.

Bug: none
Change-Id: I8182b2ab43f0c330ebdf077e9f7cbc79247da90e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202246
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33050}
2021-01-21 07:53:57 +00:00
b03b6c8a94 Move setting of encoder bitrate allocation callback type to VideoSendStream
It turned out that the negotiated rtp header extensions are not fully known in WebRtcVideoChannel::AddSendStream.

The cl also remove the unnecessary factory for creating VideoStreamEncoder.


Bug: webrtc:12000
Change-Id: If994c8deb69f3ce4212896d3ad757dac94c6e09f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198840
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32916}
2021-01-07 09:29:05 +00:00
f86cf4c2de Add support for VideoLayersAllocation for Vp9 scv/ksvc and none scalable
VideoCodecInitializer::VideoEncoderConfigToVideoCodec is modified to always set correct frame rate, width and height on spatial layer 0 so the rest of the code does not need to differentiate between scalable/none scalable codecs.


Bug: webrtc:12000
Change-Id: I5a068b98ca2038621205f55e4024f949ab51587a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198540
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32890}
2020-12-30 16:45:03 +00:00
4190ce995b Add unit test ReportsUpdatedVideoLayersAllocationWhenResolutionChanges
This test that a new allocation is reported if the input resolution
changes.

Bug: webrtc:12000
Change-Id: Iaf8be1af62bbc8a2ca19b58f0587ceacfcfa5991
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197807
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32837}
2020-12-15 17:51:05 +00:00
cde4a9f669 Enable initial frame drop for SVC 'singlecast'
Bug: none
Change-Id: Ideda726f4f7df5e92556048a199cda06261e76b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195542
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32714}
2020-11-27 14:08:45 +00:00
84bc34841b Reset initial frame dropper if the stream changes for external reasons
External reasons here are simulcast configuration and
source resolution change.
Initial frame dropper should be enabled in these cases because the
client can request way too big resolution for available bitrate and
usual quality scaling would take too long.

Bug: none
Change-Id: I02fbbd3c15b53b39672c083c2a1f9a780256c507
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195004
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32707}
2020-11-26 17:39:45 +00:00
17b29b9121 test::CreateVideoStreams: Use default unconfigured VideoStream if layer is missing in config.
Configure framerate/temporal layers via VideoEncoderConfig in VideoStreamEncoderTest..

Bug: none
Change-Id: I1104da5e576fa25746f2f2f5eaa336cd17c0093a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187488
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32500}
2020-10-27 08:19:57 +00:00
7c85d395d7 Delete unneeded includes of system_wrappers/include/sleep.h
Non-test usage is in modules/audio_device and modules/desktop_capture.

Bug: None
Change-Id: Ie7dd89aa40e6dcfa9e49e1956b87b50fd9f1c227
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190140
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32484}
2020-10-26 09:55:26 +00:00
9d69cbeabf Changes default pacing factor to 1.1x
This changes the default behavior to use pacing factor of 1.1x instead
of 2.5x, it also sets libvpx rate controler as trusted, turns on the
encoder pushback mechanism and sets spatial hysteresis to 1.2.
The unused "dynamic rate" settings in libvpx is removed.

The new settings matches what has been used in chromium since 2019.
If needed, the legacy behavior can be enabled using the field trial
WebRTC-VideoRateControl.

Bug: webrtc:10155
Change-Id: I8186b491aa5bef61e8f568e96c980ca68f0c208f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186661
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32477}
2020-10-23 13:43:32 +00:00
a94348440b VideoStreamEncoder report VideoLayersAllocation for simulcast
Adds support for Vp8 simulcast.

Bug: webrtc:12000
Change-Id: Ib24fd0542642b023ec35f7a7bdc4880d72365edf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187341
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32416}
2020-10-15 17:39:06 +00:00
b556b08668 Allow encoders to receive preferred pixel formats from native buffers
Adds a field to EncoderInfo called preferred_pixel_formats which a
software encoder populates with the pixel formats it supports. When a
kNative frame is received for encoding, the VideoStreamEncoder will
first try to get a frame that is accessible by the software encoder in
that pixel format from the kNative frame. If this fails it will fallback
to converting the frame using ToI420.

This minimizes the number of conversions made in the case that the
encoder supports the pixel format of the native buffer or where
conversion can be accelerated. For example, in Chromium, the capturer can
emit an NV12 frame, which can be consumed by libvpx which supports NV12.

Testing: Tested in Chrome with media::VideoFrame adapters.

Bug: webrtc:11977
Change-Id: I9becc4100136b0c0128f4fa06dedf9ee4dc62f37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187121
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32353}
2020-10-08 13:47:50 +00:00
1710706db8 Configure framerate/temporal layers via VideoEncoderConfig.
Bug: none
Change-Id: Ia8d350a8a71ce21691eba38db71248c10a7728a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186663
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32350}
2020-10-08 09:11:08 +00:00
dcef6410b3 Stop using VideoBitrateAllocationObserver in VideoStreamEncoder.
VideoBitrateAllocation is instead reported through the EncoderSink.
Enable VideoBitrateAllocation reporting from WebRtcVideoChannel::AddSendStream in preparation for
using the extension RtpVideoLayersAllocationExtension instead of RTCP XR.

Bug: webrtc:12000
Change-Id: I5ea8e4f237a1c4e84a89cbfd97ac4353d4c2984f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186940
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32347}
2020-10-07 18:01:13 +00:00
d0a8f51ef7 Reland Refactor reporting of VideoBitrateAllocation
Original description
 Move reporting of target bitrate to just after the encoder has been
 updated. Originall submitted as refs/heads/master@{#32275}

Patch 1 contains the original cl
,patch 2 the fix to send rtcp even if BWE does not change.

Bug: webrtc:12000
Change-Id: I16766e08229fe1f6f65f449e0e074bed03338693
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186948
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32340}
2020-10-07 11:06:56 +00:00