388 Commits

Author SHA1 Message Date
Palak Agarwal
a09f21b207 Introduce capture_time_identifier in webrtc::EncodedImage
This CL propagates capture_time_identifier introduced in
webrtc::VideoFrame and propagates it to EncodedImage. For use cases
involving EncodedTransforms, this identifier is further propagated to
TransformableVideoSenderFrame.

VideoEncoder::Encode function is overriden by each encoder. Each of
these overriden functions needs to be changed so that they can handle
this new identifier and propagate its value in the created EncodedImage.

Change-Id: I5bea4c5a3fe714f1198e497a4bcb5fd059afe516
Bug: webrtc:14878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291800
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#39374}
2023-02-22 17:08:53 +00:00
Henrik Boström
c5a4c938bb Reland "Make SimulcastIndex() and SpatialIndex() distinct (remove fallback)."
This is a reland of commit 8ad4924936dea2bd97990b0a951df93f7526f0ff

See diff between latest Patch Set and PS1. Fixes include:
- VideoStreamEncoder's call to bitrate_adjuster_->OnEncodedFrame()
  is updated to take stream index (spatial or simulcast index) instead
  of only looking at SpatialIndex().
- Migrate test-only helpers to use Spatial/SimulcastIndex correctly.

The fixes are to migrate
some test-only helpers that we had forgot to fix that are used by
external tests.

Original change's description:
> Make SimulcastIndex() and SpatialIndex() distinct (remove fallback).
>
> This CL removes the fallback logic to return the other index when the
> one requested has not been set. This means we can remove the codec gates
> that was previously needed because SpatialIndex() had multiple meanings,
> resolving the TODOs previously added in
> https://webrtc-review.googlesource.com/c/src/+/293343.
>
> We have already migrated all known external dependencies from
> SpatialIndex() to SimulcastIndex() where necessary, unblocking this CL.
>
> PSA: https://groups.google.com/g/discuss-webrtc/c/SDAVg6xJ3gY
>
> Bug: webrtc:14884
> Change-Id: I82787505ab10be151e5f64965b270c45465d63a9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293740
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39343}

Bug: webrtc:14884
Change-Id: Ib966924efca1a040dae881599f0789a7f2ab24a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294284
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39358}
2023-02-21 18:30:35 +00:00
Erik Språng
573f546197 Fix max bitrate not being resptected with some HW codecs.
This makes the max bitrate configured in RTP parameter to still be
respected, instead of being overridden by defaults in the case that an
encoder with untrusted QP is reconfigured (e.g. due to some adaptation).

Bug: webrtc:14914
Change-Id: I2d3ff645c069b80ec2e36887e6ce0ecd09a7ecbf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293944
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39357}
2023-02-21 13:57:53 +00:00
Henrik Boström
fbd0ddb32e Introduce WebRTC-VideoEncoderSettings/encoder_thread_limit:X.
As requested by a CEF hosted application (https://crbug.com/1406331)
who want to be able to limit the number of threads in a controlled
environment, this CL adds a flag to control the max limit per encoder.

For plumbing-reasons, this is placed in VideoEncoder::Settings but
with a note that this is considered an experimental API with limited
support. For now only LibvpxVp8Encoder uses it and there are no plans
to roll this out.

I have manually confirmed this is working with printf debugging,
--force-fieldtrials=WebRTC-VideoEncoderSettings/encoder_thread_limit:2
and https://jsfiddle.net/henbos/2bd6m7Lt/

Bug: chromium:1406331
Change-Id: Ib02bd83e2071034874843d3aaa0d3b0adc5bbf46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293960
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39349}
2023-02-20 14:01:32 +00:00
Wan-Teh Chang
4dac78a02b Declare 2 VideoEncoder::EncoderInfo vars as const
Declare two VideoEncoder::EncoderInfo local variables as const. This
makes it clear that they are equal to the return value of
encoder_->GetEncoderInfo().

Bug: None
Change-Id: I08ab34e670e6eb1cb3c67a48b4e9826902d9d9fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293385
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Cr-Commit-Position: refs/heads/main@{#39321}
2023-02-15 19:05:23 +00:00
Henrik Boström
2e540a28c0 Introduce EncodedImage.SimulcastIndex().
As part of go/unblocking-vp9-simulcast (Step 1), EncodedImage is being
upgraded to be able to differentiate between what is a simulcast index
and what is a spatial index.

In order not to break existing code assuming that "if codec != VP9,
SpatialIndex() is the simulcast index", SimulcastIndex() has fallback
logic to return the value of spatial_index_ in the event that
SetSimulcastIndex() has not been called. This allows migrating external
code from (Set)SpatialIndex() to (Set)SimulcastIndex(). During this
intermediate time, codec gates are still necessary in some places of
the code, see TODOs added.

In a follow-up CL, after having fixed dependencies, we'll be able to
remove the fallback logic and rely on SimulcastIndex() and
SpatialIndex() actually being the advertised index and "if codec..."
hacks will be a thing of the past!

Bug: webrtc:14884
Change-Id: I70095c091d0ce2336640451150888a3c3841df80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293343
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39318}
2023-02-15 15:02:57 +00:00
Erik Språng
17043b89ae Make sure VP9 encoders are reconfigured on layer activation.
When disabling a spatial layer, reconfiguration of the encoder is not
necessary (bitrate will never be assigned to the inactive layer anway).
This CL however makes sure we reconfigure the encoder when a spatial
layer is activated. Some encoder implementations may encoder the wrong
number of spatial layers if the active layers have not beens set
correctly.

Bug: webrtc:14809, b/261097903
Change-Id: I8d34aaec95eb50a9717c06ea38f25088e5a96429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290560
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38999}
2023-01-04 15:24:57 +00:00
Florent Castelli
acabb3641b pc: Add asynchronous RtpSender::SetParameters() call
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.

Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
2022-11-15 15:31:40 +00:00
Philipp Hancke
a1b4eb2196 generateKeyFrame: add rids argument
and do the resolution of rids to layers. This has no effect yet
since the simulcast encoder adapter (SimulcastEncoderAdapter::Encode), the VP8 encoder (LibvpxVp8Encoder::Encode) and the OpenH264 encoder (H264EncoderImpl::Encode) all generate a key frame for all layers whenever a key frame is requested on one layer.

BUG=chromium:1354101

Change-Id: I13f5f1bf136839a68942b0f6bf4f2d5890415250
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280945
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38565}
2022-11-07 15:47:51 +00:00
Evan Shrubsole
09da10e24f Add powerEfficientDecoder and powerEfficientEncoder stats
The spec for these are at https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-powerefficientdecoder and https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-powerefficientdecoder

These stats are based on the is_hardware_accelerated boolean in both the
DecoderInfo and EncoderInfo structs.

Bug: webrtc:14483
Change-Id: I4610da3c6ae977f5853a3b3424d91d864fe72592
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274409
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38441}
2022-10-19 13:15:31 +00:00
Jonas Oreland
37132e10fd RtpEncodingParameters::request_resolution patch 3
This cl/ adds resource adapation to the requested_resolution
feature. The restrictions that are sent to the video source
are also saved inside video_stream_encoder and used when
determining layer resolution.

Anticipated further patches
4) Let VideoSource do adaption if possible

Bug: webrtc:14451
Change-Id: Ia9b990a6b92b76af7ff6665a562f84585f79c35b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277580
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38306}
2022-10-06 10:29:31 +00:00
Jonas Oreland
80c87d7151 RtpEncodingParameters::request_resolution patch 2
This cl/ implements configuring of encode resolution
in the video_stream_encoder (webrtc_video_engine) in
a way that is independent of frame resolution (i.e
not using scale_resolution_down_by).

The cl/ reuses the VideoAdapter as is, and hence
the output resolution will be the same as it is today.

Anticipated further patches
3) Hook up resource adaptation
4) Let VideoSource do adaption if possible

Bug: webrtc:14451
Change-Id: I881b031c5b23be26cacfe138730154f1cb1b66a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276742
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38245}
2022-09-29 14:10:44 +00:00
Jonas Oreland
7252348d76 Create EncoderStreamFactory in VideoStreamEncoder
This cl/ changes so that the EncoderStreamFactory is
not created inside WebRtcVideoSendStream (webrtc_video_engine).

The benifit of this is that the VideoStreamEncoder can then
amend the EncoderStreamFactory with state (and types)
w/o exposing it in VideoEncoderConfig.

I.e as an alternative to changes done inside
https://webrtc-review.googlesource.com/c/src/+/276742.

The fake_webrtc_call is modified to (if needed) create
it's own EncoderStreamFactory if needed.

Note: this cl/ will have to be merged with with
https://webrtc-review.googlesource.com/c/src/+/277002.

Bug: webrtc:14451
Change-Id: I3d896b227d39725ba6409622e8d09d14bd45d5fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277160
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38237}
2022-09-28 17:47:52 +00:00
Jonas Oreland
0deda15c96 Reland "RtpEncodingParameters::request_resolution patch 1"
This reverts commit b625101da8d798c936cfd695505a5514644158b0.

Reason for revert: Found problem that was specific how
configuration is handled for VP9. A 1-line change in webrtc_video_engine.cc line 3715.
Thanks Rasmus and great that this was tested!

Original change's description:
> Revert "RtpEncodingParameters::request_resolution patch 1"
>
> This reverts commit ef7359e679e579ccb79afacf5c42e8c6020124e2.
>
> Reason for revert: Breaks downstream test
>
> Original change's description:
> > RtpEncodingParameters::request_resolution patch 1
> >
> > This patch adds RtpEncodingParameters::request_resolution
> > with documentation and plumming. No behaviour is changed yet.
> >
> > Bug: webrtc:14451
> > Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#38172}
>
> Bug: webrtc:14451
> Change-Id: I4b9590e23ec38e9e1c2e51a4600ef96b129439f2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276541
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38176}

Bug: webrtc:14451
Change-Id: Ica9b74180bce22d09bf289126bb5ac137bf9eb70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276543
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38178}
2022-09-23 11:48:19 +00:00
Björn Terelius
b625101da8 Revert "RtpEncodingParameters::request_resolution patch 1"
This reverts commit ef7359e679e579ccb79afacf5c42e8c6020124e2.

Reason for revert: Breaks downstream test

Original change's description:
> RtpEncodingParameters::request_resolution patch 1
>
> This patch adds RtpEncodingParameters::request_resolution
> with documentation and plumming. No behaviour is changed yet.
>
> Bug: webrtc:14451
> Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38172}

Bug: webrtc:14451
Change-Id: I4b9590e23ec38e9e1c2e51a4600ef96b129439f2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276541
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38176}
2022-09-23 08:27:47 +00:00
Jonas Oreland
ef7359e679 RtpEncodingParameters::request_resolution patch 1
This patch adds RtpEncodingParameters::request_resolution
with documentation and plumming. No behaviour is changed yet.

Bug: webrtc:14451
Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38172}
2022-09-22 14:16:20 +00:00
Markus Handell
e1a198b41d VideoStreamEncoder: set at target quality based on codec.
The Chromium RTCVideoEncoder unfortunately doesn't set if the
result is at target quality, and the definition of the threshold
is buried in libvpx_vp8_encoder.h.

This change
* Updates VideoStreamEncoder to postprocess an incoming EncodedImage
by interpreting the incoming QP information instead.
* Updates the related VideoStreamEncoder test to simulate an encoder
producing images around the QP threshold.
* Updates the steady state VP8 screencast QP threshold to a central
include file.
* Moves this and previously existing EncodedImage post-processing to a
new method AugmentEncodedImage.

Bug: b/245029833
Change-Id: I69ae29ffe501e84f28908f7d9a8cfd066ba82b43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38091}
2022-09-15 12:15:17 +00:00
Markus Handell
eab4ab6637 VideoStreamEncoder: add Encode and QP feedback logging.
Bug: b/245029833
Change-Id: I02f867d2505ad5748bda3c72fd690b2c07647732
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275202
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38064}
2022-09-12 20:06:37 +00:00
Sergey Silkin
ceb71cd557 Switch encoder on any critical frame encode error (returncode < 0)
Before this change encoder switch was triggered only if encode() returns WEBRTC_VIDEO_CODEC_ENCODER_FAILURE. Android HW encoder wrapper never return this error code. It returns WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE [1, 2] or WEBRTC_VIDEO_CODEC_ERROR [3].

Change value of WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT from -14 to 5 to avoid it to be interpreted as a critical error.

[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/android/src/jni/video_encoder_wrapper.cc;drc=c05a1be5b49f5c03b6955b05bcbf47609e1b0381;l=324

[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/android/src/jni/video_encoder_wrapper.cc;drc=c05a1be5b49f5c03b6955b05bcbf47609e1b0381;l=335

[3] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/android/src/jni/video_encoder_wrapper.cc;drc=c05a1be5b49f5c03b6955b05bcbf47609e1b0381;l=331

Bug: b/243402636
Change-Id: Iaf0129f3f9d71c07bb06804fe1f92a1f84f6da26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274402
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38060}
2022-09-12 12:43:17 +00:00
Danil Chapovalov
9e09a1f327 Replace Thread::Invoke with Thread::BlockingCall
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed

Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
2022-09-09 10:44:17 +00:00
Erik Språng
5e13d0599c Request refresh frame after unpausing encoder with native frame drop.
If a "normal" software buffer frame is dropped during paused state, we
store it as a pending frame and try encoding it after the pause state is
lifted. However, native frames are dropped entirely since keeping e.g.
texture handles for long time periods can lead to side effects.

Work around this by requesting a refresh frame after unpausing if the
dropped frame flag is set.

Bug: webrtc:14276
Change-Id: I9edd1e99454e082bcfe29f3d9041026dd8a390d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270220
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37660}
2022-08-02 11:35:16 +00:00
Danil Chapovalov
c05a1be5b4 Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
Bug: webrtc:14245
Change-Id: I8de2c23da5fbdfc0b1efbbe07fb6e8de744424a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268191
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37565}
2022-07-20 08:15:08 +00:00
Danil Chapovalov
03f8b8a241 In video replace non-owning pointer to rtc::TaskQueue with non-owning pointer to TaskQueueBase
rtc::TaskQueue is a simple wrapper over TaskQueueBase and adds no
extra features when task queue is used without passing ownership.

Reducing usage of the internal rtc::TaskQueue wrapper gives users more flexibility how TaskQueueBase* is stored.

Bug: webrtc:14169
Change-Id: If5c8827544c843502c7dfcef775ac558de79ec3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268189
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37549}
2022-07-18 13:59:32 +00:00
Danil Chapovalov
95eeaa7aca Migrate video/ to absl::AnyInvocable based TaskQueueBase interface
Bug: webrtc:14245
Change-Id: Ibd98d3a0c548443578953ef3e25aee9919eea3d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267980
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37465}
2022-07-06 14:40:25 +00:00
Niels Möller
b5b159d98c Update old TODO comments
Bug: None
Change-Id: I531ed648fe3d1f0dd1202f53c59ed023aed1ea7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267664
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37432}
2022-07-05 09:09:44 +00:00
Artem Titov
c374d11fac Move to_queued_task.h and pending_task_safety_flag.h into public API
Bug: b/235812579
Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-17 09:20:39 +00:00
Jonas Oreland
6545516a14 RtpSenderInterface::SetEncoderSelector
This cl/ adds a way of setting an EncoderSelector on a specific
RtpSenderInterface. This makes it possible to easily use different
EncoderSelector on different streams within the same or different PeerConnections.

The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor.

Iff a EncoderSelector is set on the RtpSender, it will take precedence
over the VideoEncoderFactory::GetEncoderSelector.

Bug: webrtc:14122
Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37150}
2022-06-08 11:06:52 +00:00
Tommi
4d6dd17153 Change VideoStreamEncoder::Stop() to handle stopped TQ
In case the encoder TQ has been stopped and doesn't accept more tasks,
we could end up in a hung state during Stop(). This is a hypothetical
situation, but can be simulated in a test and avoided.

Bug: webrtc:14063
Change-Id: I20f48b11b6266f6875ed5e69de3529212505e439
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258125
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36964}
2022-05-23 08:34:36 +00:00
Asa Persson
2ee3e4db22 Use resolution bitrate limits if one spatial layer is configured via scalability mode.
Bug: webrtc:13960
Change-Id: Ie9238f3352b0d9d92fda97a250de0792e6bbfc9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261721
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36948}
2022-05-20 13:25:51 +00:00
Niels Möller
807328fec7 Move frame drop config to VideoCodec and VideoEncoderConfig.
Intend to delete corresponding codec-specific settings in a followup.

Bug: webrtc:6883
Change-Id: I78ab07729a5aee1055f80d39d8f7289beb6721e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262244
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36882}
2022-05-13 13:40:14 +00:00
Niels Möller
79d566b0cf New enum ScalabilityMode.
Used instead of string representation in lower-levels of encoder configuration, to avoid string comparisons (with risk of misspelling) in lots of places.

Bug: webrtc:11607
Change-Id: I4d51c2265aac297c29976d2aa601d8ffb33b7326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259870
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36706}
2022-04-29 12:16:42 +00:00
philipel
6daa3048fc Added OnResolutionChange to EncoderSelectorInterface.
Bug: webrtc:12406
Change-Id: I0160636d93ad0a33caf7ae7443cefe321a191406
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258442
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36517}
2022-04-11 10:04:57 +00:00
Byoungchan Lee
13fe3674ff Fix to not crash when VideoEncoderFactory fails to create encoder
Bug: webrtc:13082
Change-Id: I5f1cfa7db6259e71ce3fc18281a3d084c32911ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257923
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#36501}
2022-04-08 15:52:44 +00:00
Erik Språng
e4589cb55e Reduce libvpx VP9 complexity setting on <= 2 core machines.
This CL sets speed 9 for all resolutions when two or less cores are
available, as a heuristic for a "slow" machine.
This gives a large speed bost at a relatively small quality loss.

A field-trial kill-switch is available to override this behavior.

Bug: webrtc:13888
Change-Id: I24278a45de000ad7984d0525c47d9eb6b9ab6b60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257421
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36466}
2022-04-06 16:08:00 +00:00
Jonas Oreland
e62c2f2c77 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
rename WebRtcKeyValueConfig to FieldTrialsView

Bug: webrtc:10335
Change-Id: If725bd498c4c3daf144bee638230fa089fdde833
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 10:14:00 +00:00
Jonas Oreland
8ca06137dc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 4/inf
convert almost all of video/ (and the collateral)

Bug: webrtc:10335
Change-Id: Ic94e05937f54d11ee8a635b6b66fd146962d9f11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36192}
2022-03-14 14:36:35 +00:00
Jonas Oreland
c7f691a71a WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
convert call/ (and the collaterals)

Bug: webrtc:10335
Change-Id: I8f6bc13c032713aa2a947724b464f6f35454d39a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254320
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36165}
2022-03-09 22:17:52 +00:00
Artem Titov
6cae2d5513 Reland "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely"
This reverts commit 3f87250a4f0e6c69002fbcdfb995b0dfcd7bf710.

Reason for revert: Downstream is fixed

Original change's description:
> Revert "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely"
>
> This reverts commit 5f0eb93d2a44cec2102fc8c3757d5bb814bd145f.
>
> Reason for revert: Breaks downstream project. I'm going to fix that one and create a reland of this CL after.
>
> Original change's description:
> > Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely
> >
> > Bug: webrtc:13555, webrtc:13082
> > Change-Id: Iff2cda6f516739419e97e975e03f77a98f74be03
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249260
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Artem Titov <titovartem@webrtc.org>
> > Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
> > Cr-Commit-Position: refs/heads/main@{#35805}
>
> TBR=hta@webrtc.org,titovartem@webrtc.org,daniel.l@hpcnt.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I33d497f1132adfe6d151023195a388d9b7d548f9
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:13555, webrtc:13082
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249364
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Owners-Override: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35807}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:13555, webrtc:13082
Change-Id: I7ef1ef3b6e3c41b1a96014aa75f003c0fcf33949
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249365
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35814}
2022-01-27 12:55:44 +00:00
Artem Titov
3f87250a4f Revert "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely"
This reverts commit 5f0eb93d2a44cec2102fc8c3757d5bb814bd145f.

Reason for revert: Breaks downstream project. I'm going to fix that one and create a reland of this CL after.

Original change's description:
> Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely
>
> Bug: webrtc:13555, webrtc:13082
> Change-Id: Iff2cda6f516739419e97e975e03f77a98f74be03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249260
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
> Cr-Commit-Position: refs/heads/main@{#35805}

TBR=hta@webrtc.org,titovartem@webrtc.org,daniel.l@hpcnt.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I33d497f1132adfe6d151023195a388d9b7d548f9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:13555, webrtc:13082
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249364
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35807}
2022-01-26 14:56:14 +00:00
Byoungchan Lee
5f0eb93d2a Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely
Bug: webrtc:13555, webrtc:13082
Change-Id: Iff2cda6f516739419e97e975e03f77a98f74be03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249260
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35805}
2022-01-26 14:22:16 +00:00
Tommi
62b01db428 VideoStreamEncoder - wait less.
Some of the state that's managed in VideoStreamEncoder, is updated
and accessed on the encoder queue, but a method that's used for testing
only (GetAdaptationResources()), represents a race between PostTask
operations that update state and checking for said state on the worker
thread.

This CL removes Wait() operations related to adaptation resources from
the common path and puts one in the test path instead.

Bug: webrtc:13612
Change-Id: Ie3e018e815e24951bc0634ed70de17eaf336a508
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249220
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35797}
2022-01-26 09:53:33 +00:00
Sergey Silkin
e1cd3ad4f5 Switch encoder on init failure
Currently if encoder initialization fails WebRTC doesn't send any video.
This CL adds functionality that changes encoder type in such case and
restores the video. If encoder selector is available we switch to
encoder it recommends. Otherwise, VP8 is used as the default fallback
encoder.

Bug: webrtc:13572
Change-Id: Ifcdf707a575711f5ff81f9451caf30140c9171dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246960
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35761}
2022-01-21 12:05:17 +00:00
Markus Handell
818e7fbc64 ZeroHertzAdapterMode: handle pending key frame requests.
The frame cadence adapter ignores key frame processing that happens
before the point where zero-hertz mode is activated, which leads to
no refresh frame requests if the key frame request comes too early.

Fix this to register a pending refresh frame request that gets
serviced when zero-hertz mode is activated. The CL changes the
FrameCadenceAdapterInterface::ProcessKeyFrameRequest from returning
whether to request a refresh frame into the frame cadence adapter
actively doing so itself via a new Callback::RequestRefreshFrame
API.

go/rtc-0hz-present

Bug: chromium:1255737
Change-Id: I53c2dbf6468e883eb2a2e81498e7134b1b35c336
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242963
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35598}
2021-12-30 13:37:10 +00:00
Markus Handell
e59fee87fb ZeroHertzAdapterMode: activate earlier.
Careful analysis of logs related to the requesting of refresh
frames from the source revealed an uncomfortable truth:
zero-hertz mode activates first when the first frame has been
received in the VideoStreamEncoder, because the number of simulcast
layers can only be computed when frame dimensions are known. This
fact means that the currently implemented logic for requesting
refresh frames is noneffective.

Fix this by
1. Activating zero-hertz mode prior of knowing the final layer
count. This causes refresh frame requests to happen without any
frames received from the source.
2. Making layer count dynamically configurable. Zero-hertz mode
considers layer quality unconverged after such a reconfiguration.

go/rtc-0hz-present

Bug: chromium:1255737
Change-Id: I0ecea4d2a8442a00e3b79b146dd39a5f4ac561d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35593}
2021-12-29 20:28:38 +00:00
Markus Handell
2e0f4f0f37 ZeroHertzAdapterMode: handle key frame requests.
Under zero-hertz mode, provided that a frame arrived to the
VideoStreamEncoder, the receiver may experience up to a second
between incoming frames. This results in key frame requests getting
serviced with that delay, which is undesired.

What's worse is also the fact that if no frame ever arrived to the
VideoStreamEncoder, it will not service the keyframe requests at all
until the first frame comes.

This change introduces VideoSourceInterface::RequestRefreshFrame
which results in a refresh frame being sent from complying sources.
The method is used under zero-hertz mode from the VideoStreamEncoder
when frames didn't arrive to it yet (with changes to the zero-hertz
adapter).

With this change, when the frame adapter has received at least one
frame, it will conditionally repeat the last frame in response to the
key frame request.

go/rtc-0hz-present

Bug: chromium:1255737
Change-Id: I6f97813b3a938747357d45e5dda54f759129b44d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242361
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35562}
2021-12-21 19:52:56 +00:00
Markus Handell
8d87c463d9 ZeroHertzAdapterMode: slow down repeats on quality convergence.
The frame cadence adapter previously resulted in unconditional
frame repeating at max FPS. Change this to slow down to an idle
rate (1 Hz) when quality convergence in all configured spatial
layers has been achieved.

go/rtc-0hz-present

Bug: chromium:1255737
Change-Id: Ifa593dbf8a61aa29da20ac250da332734ae82791
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241421
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35547}
2021-12-16 12:01:30 +00:00
Markus Handell
ee22543829 Zero-hertz encoding mode: avoid encoder bitrate overshooting.
The encoders wrapped in VideoStreamEncoder grossly over-estimates
available bitrate when capture FPS falls close to zero, and frames
re-commence highly frequent delivery. Avoid this by moving the input
RateStatistics inside VSE into the frame cadence adapter, and changing
the reported framerate under zero-hertz encoding mode to always return
the configured max FPS.

Bug: chromium:1255737
Change-Id: Iaa71ef51c0755b12e24e435d86d9562122ed494e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239126
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35431}
2021-11-29 13:48:00 +00:00
Markus Handell
d7eef66a39 VideoStreamEncoder: move PostTasks to WebRtcVideoChannel.
This change moves the responsibility of posting
EncoderSwitchRequestCallback calls closer to the top-level
users which has a better idea about threading requirements.

The change is planned to be followed-up with more changes removing
the need for VSE to post to the worker thread.

Bug: webrtc:13414, chromium:1255737
Change-Id: I57a2962a70e9f245460c59c0d61824371394b952
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238420
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35387}
2021-11-19 11:25:11 +00:00
Markus Handell
9a478b527d VideoStreamEncoder: expect frame entry on the encoder queue.
This change switches the sequence used by the FrameCadenceAdapter
to be the encoder_queue, enabling VideoStreamEncoder::OnFrame to be
invoked directly on the encoder_queue and eliminates the contained
PostTasks.

Bug: chromium:1255737
Change-Id: Ib86fc96ad2be9a38585fef2535855e3f9cc7e57c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238171
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35380}
2021-11-18 15:35:17 +00:00
Niels Möller
13d163654a Delete support for has_internal_source
Bug: webrtc:12875
Change-Id: I9683e71e1fe5b24802033ffcb32a531ca685fc6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179220
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35353}
2021-11-16 11:29:40 +00:00