1554 Commits

Author SHA1 Message Date
philipel
ca18809ee5 Move RtpFrameObject and EncodedFrame out of video_coding namespace.
Bug: webrtc:12579
Change-Id: Ib7ecd624eb5c54abb77fe08440a014aa1e963865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33542}
2021-03-23 14:22:47 +00:00
philipel
6a6715042a Move RtpFrameReferenceFinder out of video_coding namespace.
Namespace used because of copy-pasting an old pattern, should never have been used in the first place. Removing it now to make followup refactoring prettier.

Bug: webrtc:12579
Change-Id: I00a80958401cfa368769dc0a1d8bbdd76aaa4ef5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212603
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33536}
2021-03-23 08:48:37 +00:00
Henrik Boström
f412976eca Provide a default implementation of NV12BufferInterface::CropAndScale.
This avoids falling back on the VideoFrameBuffer::CropAndScale default
implementation which performs ToI420. This has two major benefits:
1. We save CPU by not converting to I420 for NV12 frames.
2. We make is possible for simulcast encoders to use Scale() and be
   able to trust that the scaled simulcast layers have the same pixel
   format as the top layer, which is required by libvpx.

In order to invoke NV12Buffer::CropAndScaleFrom() without introducing a
circular dependency, nv12_buffer.[h/cc] is moved to the "video_frame"
build target.

Bug: webrtc:12595, webrtc:12469
Change-Id: I81aac5c6b3e81c49f32a7be6dc2640e6b40f7692
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212643
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33521}
2021-03-22 11:09:36 +00:00
Niels Möller
490c1503d9 Delete unowned buffer in EncodedImage.
Bug: webrtc:9378
Change-Id: Ice48020c0f14905cbc185b52c88bbb9ac3bb4c93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128575
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33510}
2021-03-19 14:12:28 +00:00
Sergey Silkin
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
Sergey Silkin
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
Alessio Bazzica
bc1c93dc6e Add remote-outbound stats for audio streams
Add missing members needed to surface `RTCRemoteOutboundRtpStreamStats`
via `ChannelReceive::GetRTCPStatistics()` - i.e., audio streams.

`GetSenderReportStats()` is added to both `ModuleRtpRtcpImpl` and
`ModuleRtpRtcpImpl2` and used by `ChannelReceive::GetRTCPStatistics()`.

Bug: webrtc:12529
Change-Id: Ia8f5dfe2e4cfc43e3ddd28f2f1149f5c00f9269d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211041
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33452}
2021-03-12 20:39:50 +00:00
philipel
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
philipel
7c7885c016 Remove NTP timestamp from PacketBuffer::Packet.
Bug: webrtc:12579
Change-Id: I64ca0ddb6f5c20bef5e9503955e0e4b4c484a1e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211662
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33448}
2021-03-12 15:19:35 +00:00
Sergey Silkin
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
Ilya Nikolaevskiy
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
Sergey Silkin
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
Sergey Silkin
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
Di Wu
668dbf66ce [Stats] Populate "frames" stats for video source.
Spec: https://www.w3.org/TR/webrtc-stats/#dom-rtcvideosourcestats-frames

Wiring up the "frames" stats with the cumulative fps counter on the video source.

Tests:
./out/Default/peerconnection_unittests
./out/Default/video_engine_tests

Bug: webrtc:12499
Change-Id: I4103f56ed04cb464f5f7e70fbf2d77c25a867a68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208782
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33404}
2021-03-09 08:54:38 +00:00
Guido Urdaneta
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
Sergey Silkin
964a88fce1 Prevent possible out-of-bounds access
It read from simulcast_layers[0] even if vector is empty.

Bug: none
Change-Id: I293890feda70022beae4247ab10cf8182b4cf4a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209706
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33391}
2021-03-05 17:48:48 +00:00
Åsa Persson
4a4273bf05 VP9 ResolutionBitrateLimits: If bitrates are configured, use intersection.
Bug: none
Change-Id: I58ada41d7a196837b35df4cf61f7e7561998cf13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209704
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33390}
2021-03-05 17:39:04 +00:00
Sergey Silkin
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
Sergey Silkin
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
Björn Terelius
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
Bjorn Terelius
83e6eceac7 Comment out uninstantiated parametrized PC full stack test
Bug: None
Change-Id: If4756fd30df5788fdbe8bfcb36f5333167c50669
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209460
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33376}
2021-03-03 17:31:46 +00:00
Sergey Silkin
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
Åsa Persson
64f7da0b16 Use input_state to get pixels for single active stream.
Bug: none
Change-Id: I103d2cc111ca08d1e5acde1f25c125c075502eca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208226
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33358}
2021-03-01 15:58:03 +00:00
Åsa Persson
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
Henrik Boström
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
Åsa Persson
1dd94a023a Use pixels from single active stream if set in CanDecreaseResolutionTo
Simulcast with one active stream:
Use pixels from single active stream if set (instead of input stream which could be larger) to avoid going below the min_pixel_per_frame limit when downgrading resolution.

Bug: none
Change-Id: I65acb12cc53e46f726ccb5bfab8ce08ff0c4cf78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208101
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33309}
2021-02-22 10:25:32 +00:00
Danil Chapovalov
1fbff10254 In RtpVideoStreamReceiver change way to track time for the last received packet.
Instead of tracking packets accepted by PacketBuffer, track all incoming
packets, including packets discarded before getting into PacketBuffer.

Bug: b/179759126
Change-Id: I4d270c61455608fb78b0df8f27760868f4c98205
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208289
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33305}
2021-02-19 17:26:54 +00:00
Johannes Kron
16359f65c4 Delay creation of decoders until they are needed
Before this CL, WebRTC created a decoder for each negotiated codec
profile. This quickly consumed all available HW decoder resources
on some platforms. This CL adds a field trial,
WebRTC-PreStreamDecoders, that makes it possible to set how many
decoders that should be created up front, from 0 to ALL. If the
field trial is set to 1, we only create a decoder for the
preferred codec. The other decoders are only created when they are
needed (i.e., if we receive the corresponding payload type).

Bug: webrtc:12462
Change-Id: I087571b540f6796d32d34923f9c7f8e89b0959c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208284
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33300}
2021-02-19 12:08:49 +00:00
Åsa Persson
e927c0ff3e QualityScalingTests: Move encoder factory creation to ScalingObserver.
Bug: none
Change-Id: I44131952c8ef8efa62049702ae1c715a7c419dd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208102
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33290}
2021-02-17 14:27:55 +00:00
Åsa Persson
2c8d9299c8 QualityScalingTests: Add tests for VP9.
Bug: none
Change-Id: Ic6e8539dfd1a43581eb4bfc26a2b04b9cd6a4cab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207435
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33277}
2021-02-16 09:53:44 +00:00
Johannes Kron
b73c9f0bc3 Extract SystemTimeNanos to its own file
Prepare for this function to be overridden when WebRTC is included
into other applications such as chromium. This will make it
possible to remove code that keeps track of the difference between
WebRTC and chromium time.

Bug: chromium:516700
Change-Id: I73133804f945cc439f9827ec68a8e67b96d8560f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204304
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33273}
2021-02-15 22:38:46 +00:00
Tomas Gunnarsson
8408c9938c Remove 'secondary sink' concept from webrtc::VideoReceiveStream.
In practice, support for multiple sinks is not needed and supporting
the API that allows for dynamically adding/removing sinks at runtime,
adds to the complexity of the implementation.

This CL removes that Add/Remove methods for secondary sinks as well
as vectors of callback pointers (which were either of size 0 or 1).
Instead, an optional callback pointer is added to the config struct
for VideoReceiveStream, that an implementation can consider to be
const and there's not a need to do thread synchronization for that
pointer for every network packet.

As part of webrtc:11993, this simplifies the work towards keeping
the processing of network packets on the network thread. The secondary
sinks, currently operate on the worker thread.

Bug: webrtc:11993
Change-Id: I10c473e57d3809527a1b689f4352e903a4c78168
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207421
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33272}
2021-02-15 18:08:17 +00:00
Åsa Persson
453a125438 Remove no longer needed FrameDroppingOn setting in QualityScalingTests.
Bug: none
Change-Id: Id3868f947584616a7027a3985155a79c01e6dbb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207422
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33269}
2021-02-15 15:10:49 +00:00
philipel
9aa9b8dbbe Prepare to replace VideoLayerFrameId with int64_t.
Bug: webrtc:12206
Change-Id: I10bfdefbc95a79e0595956c1a0e688051da6d2b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207180
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33265}
2021-02-15 14:42:02 +00:00
philipel
f109193fba Remove VideoLayerFrameId::spatial_layer, use EncodedImage::SpatialIndex instead.
Next step is to replace VideoLayerFrameId with int64_t.

Bug: webrtc:12206
Change-Id: I414f491e383acf7f8efd97f7bf93dc55a5194fbf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206804
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33245}
2021-02-12 11:16:23 +00:00
Etienne Pierre-Doray
cc4743795b [Battery]: Delay start of CallStats.
To avoid unnecessary repeating tasks, CallStats' timer is started only
upon Call::EnsureStarted().

Bug: chromium:1152887
Change-Id: I1015315f42127bf510affc3d22c930b20eac8bba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206880
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33232}
2021-02-11 12:00:05 +00:00
Artem Titov
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
Ilya Nikolaevskiy
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
Åsa Persson
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
Artem Titov
c8421c4c3e Replace rtc::ThreadChecker with webrtc::SequenceChecker
Bug: webrtc:12419
Change-Id: I825c014cc1c4b1dcba5ef300409d44859e971144
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205002
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33136}
2021-02-02 14:56:27 +00:00
Andrey Logvin
e7c79fd3d6 Remove from chromium build targets that are not compatible with it.
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.

`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.

Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 13:46:19 +00:00
Ilya Nikolaevskiy
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
Åsa Persson
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
Andrey Logvin
7864600a6e Add absl_deps field for rtc_test and rtc_executable
To be able to build these targets in chromium we need to replace all abseil dependencies with "//third_party/abseil-cpp:absl".

Bug: webrtc:12404
Change-Id: Ie0f6af73f2abc73e5744520cfd9a6414e2f948e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202762
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33108}
2021-01-29 16:40:49 +00:00
Ilya Nikolaevskiy
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
Per Kjellander
1aa1d6463a Ensure VideoLayersAllocation.frame_rate_fps can not overflow
Bug: webrtc:12000
Change-Id: I14d5f0f987fb20bd74e0428b3791bf370476296e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204220
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33089}
2021-01-28 12:39:30 +00:00
Sergey Silkin
8c007fffea Restrict usage of resolution bitrate limits to singlecast
Bug: none
Change-Id: I4d0726d45a517b51eae124dc23e533910ede7cc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203262
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33061}
2021-01-22 21:41:00 +00:00
Jakob Ivarsson
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
Åsa Persson
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
Åsa Persson
a24d35eb09 AlignmentAdjuster: take reduced layers into account for default downscaling.
Bug: none
Change-Id: Id70f7763d2e1b11c24ad98774f1bf6a661728437
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202257
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33038}
2021-01-19 16:59:11 +00:00