2679 Commits

Author SHA1 Message Date
Tony Herre
6d262c504a Add TransformableVideoFrameInterface::Metadata()
Add a method to TransformableVideoFrameInterface which returns a new
instance of VideoFrameMetadata which the caller can move and use as
they like.
This will replace the existing GetMetadata which returns a dangerous const ref to a field which might change if someone calls SetMetadata
etc. That method will be deprecated as soon as we've migrated Chromium
usages.

Bug: webrtc:14708
Change-Id: Id7c15f33d6ec28c4a975ce250cdc791d7a3087bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Tove Petersson <tovep@google.com>
Cr-Commit-Position: refs/heads/main@{#39403}
2023-02-27 15:38:32 +00:00
Mirko Bonadei
b5f2c7edda Make Y4mFrameGenerator read FPS from file format.
Bug: b/269577953
Change-Id: Ied0072e1fdfbfb4d2b11e74a814c0718cad01d66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294862
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39401}
2023-02-27 14:19:04 +00:00
Markus Handell
2a256c82ec Implement support for Chrome task origin tracing. #2/4
This prepares TaskQueueBase sub classes to be able to migrate to
the location and traits-based API. It re-introduces a Location class
into the webrtc namespace, which is meant to be overridden by Chromium.

Bug: chromium:1416199
Change-Id: I712c7806a71b3b99b2a2bf95e555b357c21c15ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294381
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39400}
2023-02-27 13:17:38 +00:00
Sergey Silkin
8566e779e3 Add samples sum calculation
Bug: b/261160916, webrtc:14852
Change-Id: I88e464fce4673dd9b9683219b8d2837206579ba5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293942
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39386}
2023-02-24 11:48:39 +00:00
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
Philipp Hancke
b660b7a89c Enable multithreaded OpenH264 encoding behind field trial
This uses the field trial introduced is crbug.com/1406331 and
extends the usage to OpenH264. This simplifies experimentation
whether this change improves performance without requiring
multi-slice encoding.

BUG=webrtc:14368

Change-Id: I0031e59059f7113dd5453234869c957d46f311bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39371}
2023-02-22 14:26:33 +00:00
Palak Agarwal
b57053ec21 Use type webrtc::Timestamp as capture_time_identifier in VideoFrame
Replace the existing variable capture_time_identifier_ms_ with
capture_time_identifier_ in webrtc::VideoFrame and
webrtc::VideoFrame::Builder. This variable uses webrtc::Timestamp as its
type versus using int64_t which creates confusion about whether it is
being recorded in milliseconds/microseconds.

Change-Id: I0b83a6235fb1d5732f7afe2c14d7d6121f1e985b
Bug: webrtc:14878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#39369}
2023-02-22 13:50:07 +00:00
Philipp Hancke
fe1b39a648 stats: Deprecate RTCStatsReport(int64 timestamp_us)
in favor of the variant with (or returning) a Timestamp object.

BUG=webrtc:14813,webrtc:13756

Change-Id: I7b40f48f640a8be40a134b380a7a1b99cc99913b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294287
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39366}
2023-02-22 12:32:02 +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
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
Henrik Boström
79a6f87648 Revert "Make SimulcastIndex() and SpatialIndex() distinct (remove fallback)."
This reverts commit 8ad4924936dea2bd97990b0a951df93f7526f0ff.

Reason for revert: Breaks downstream projects

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: Ibcb834a1519930336fa50e8e9d8d0137972e28e6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294282
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39347}
2023-02-20 12:47:37 +00:00
Philipp Hancke
e5ab6c3bb0 stats: remove deprecated timestamp_us constructor variant
BUG=webrtc:14813

Change-Id: I56d28385f679b399cb2059f4c4c3d43e84a89b8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293580
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39346}
2023-02-20 12:18:42 +00:00
Henrik Boström
8ad4924936 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}
2023-02-20 10:48:24 +00:00
Mirko Bonadei
3ea7162c3c Make FrameGeneratorInterface::GetResolution pure virtual.
Bug: b/269577953
Change-Id: Ia8d370b9741fe3ed19ce276265ff7de7dcd061d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293961
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39335}
2023-02-17 17:03:24 +00:00
Mirko Bonadei
f1e392214d Make frame generators return the target resolution.
Bug: b/269577953
Change-Id: Ib3db0017becb8a6a680997f59e0f9050a42a3a79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293940
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39331}
2023-02-17 13:20:32 +00:00
Tony Herre
0277f2b4a7 Add GetDirection method to MockTransformableVideoFrame
Allow mocking of GetDirection in tests.
Required for Chromium adoption of this mock:
https://chromium-review.googlesource.com/c/chromium/src/+/4236916

Bug: chromium:1414370
Change-Id: I2e7443a1bf24966cfcfaeadf47c5b29375e84f99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293745
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39329}
2023-02-16 18:55:56 +00:00
Dor Hen
99e002fdc4 Add APIs audio encoder/decoder factories in PeerConfigurer
In Meta we have our own audio encoder/decoder factories and we would like to exercise those in the peer connection e2e test framework.
Also, looks cleaner to have the APIs for both video and audio :)

Bug: webrtc:14910
Change-Id: Ibd1e0f39fc809882ef17b3de3154fdf4b567013b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293782
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39328}
2023-02-16 15:53:01 +00:00
Sergey Silkin
72b99a1128 Test Android HW codecs
Bug: b/261160916, webrtc:14852
Change-Id: Iebeab244a9ca6ae196735016064ccd056b7c888e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293401
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39326}
2023-02-16 14:01:52 +00:00
Wan-Teh Chang
bd86684bf3 Make VideoEncoder::GetEncoderInfo() pure virtual
Bug: webrtc:9722
Change-Id: I831a9c460425be86e5da2761769b8eecf231462f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293386
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Cr-Commit-Position: refs/heads/main@{#39319}
2023-02-15 17:26:07 +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
philipel
04e9354557 Remove deprecated VideoStreamDecoderInterface and FrameBuffer2.
Bug: webrtc:14875
Change-Id: I46ea21d9ed46283ad3f6c9005ad05ec116d841f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291701
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39304}
2023-02-13 16:25:00 +00:00
Tony Herre
fd877d996f Consolidate TransformableVideoFrame mocks used inside webrtc
Also move the frame_transformer_factory_unittest build target into the
if(rtc_include_tests) block, so it's not compiled without the mock.

Bug: chromium:1414370
Change-Id: I12653b173b419ec20bfad904e24a4d965e7e7830
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292863
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39288}
2023-02-09 16:06:29 +00:00
Tony Herre
168d11cba9 Deprecate TransformableVideoFrame GetAdditionalData
It's unused in Chromium and internally - GetMetadata() provides
sufficient information.

Bug: chromium:1414370
Change-Id: Id93bdccbda85090c1aa2fabf5d6b7b79f2b1e2e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292862
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39287}
2023-02-09 16:04:10 +00:00
Wan-Teh Chang
0c4c9be436 video_encoder.h: update kFullFramerate in comment
During code review, kFullFramerate was renamed kMaxFramerateFraction but
the uses of kFullFramerate in a comment were not updated. See
https://webrtc-review.googlesource.com/c/src/+/117900/3..4

Change-Id: I6b3c06b4c5b302e8ba40bde4ba722b94aab191eb
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292801
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39285}
2023-02-09 15:17:50 +00:00
Palak Agarwal
617d89a385 Add capture time as identifier in webrtc::VideoFrame
This will be used by third_party/blink/renderer/platform/peerconnection/webrtc_video_track_source.cc to provide capture_time_identifier_ms_ from
media::VideoFrame.

This identifier would then be passed to webrtc::EncodedFrame and
webrtc::TransformableVideoSenderFrame (in the future CLs) to be used as
an identifier for encoded frames.


Bug: webrtc:14878
Change-Id: I1d8a27891323d86fdc2f014988a8da572df84119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291805
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#39270}
2023-02-08 11:05:47 +00:00
Tony Herre
b459deaf38 Add ssrc to VideoFrameMetadata used in encoded transforms
This allows callers to modify an encoded video frame's SSRC via the
setMetadata() call, which we'd like to do from Chrome, to allow using
an encoded frame from one PC on a different one.

Bug: webrtc:14709
Change-Id: Ia6b33761a3f63038f6eabbcd848916877e24454b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292380
Auto-Submit: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39266}
2023-02-08 06:34:27 +00:00
Sergey Silkin
c6ff4bc793 Do not transfer ownership of codecs to tester
Passing of ownership of codecs to tester is not strictly needed. We may need to continue using a codec after test. For example, to check codec state or to use the same codec instance in next test.

Bug: b/261160916, webrtc:14852
Change-Id: I179b262116d7de76b8171f0409f943ad6d87433e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291802
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39256}
2023-02-03 14:29:43 +00:00
Tony Herre
be9b576188 Move video video receiver transformable frame to modules/rtc_rtcp/source
Step 1 of combining the sender and receiver types

Also moved the RtpFrameObject to rtp_rtcp/source, as it's heavily used
by the transformable receiver frame, I couldn't work out a better way
of managing the dependencies, and everything else seemed to work fine.

Bug: chromium:1412687
Change-Id: I55e816a0d7aa2962560ff9ebaf30ad63ab0b9810
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291710
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39255}
2023-02-03 12:59:19 +00:00
Sergey Silkin
6c60f72a6b Refactor video codec testing stats
This CL introduces VideoCodecStats and VideoCodecStatsImpl which provide baseline functionalities for storing, slicing and aggregation of encoded and/or decoded video frame statistics. To facilitate metrics logging (not implemented yet), SamplesStatsCounter is used for stream parameters.

VideoCodecStats/VideoCodecStatsImpl will replace existing VideoCodecTestStats/VideoCodecTestStatsImpl.

Bug: b/261160916, webrtc:14852
Change-Id: I0f96ce1ed9be3aee2a702804612524676c9882fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291323
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39248}
2023-02-02 15:56:40 +00:00
Mirko Bonadei
0b7184ce06 Add possibility to set MetricsSet metadata.
Bug: b/266997275
Change-Id: I2c4fadcff7044a8c72ef7e46caf4eff398e29f91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291700
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39237}
2023-01-31 12:41:47 +00:00
Mirko Bonadei
e5922834f8 Add 'metadata' field to MetricsSet proto.
Bug: b/266997275
Change-Id: Iece033b0bd3b6e2946a57ae19dd4ff0a0417694f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291535
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39232}
2023-01-31 09:45:26 +00:00
Artem Titov
a617867a45 Reland "Migrate WebRTC documentation to new renderer"
This reverts commit 0f2ce5cc1c779f9bf33f51f29bfffbcbe105d1b1.

Reason for revert: Downstream infrastructure should be ready now

Original change's description:
> Revert "Migrate WebRTC documentation to new renderer"
>
> This reverts commit 3eceaf46695518f25bef43f155f82ed174827197.
>
> Reason for revert:
>
> Original change's description:
> > Migrate WebRTC documentation to new renderer
> >
> > Bug: b/258408932
> > Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39205}
>
> Bug: b/258408932
> Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39209}

Bug: b/258408932
Change-Id: Ia172e4a6ad1cc7953b48eed08776e9d1e44eb074
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291660
Owners-Override: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39231}
2023-01-31 09:30:04 +00:00
Tony Herre
64ce699f4b Propagate Video CSRCs modified by an insertable streams frame transform
Allow CSRCs to be modified per-frame in an Encoded Insertable Streams
transform, to support a web API which allows per-frame CSRC
modifications to signal when a JS application has changed the source
of the video which is written into an encoded frame.

Initially only for Video, with Audio support likely to follow later.

Bug: webrtc:14709
Change-Id: Ib34f35faa9cee56216b30eaae42d7e65c78bb9f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291324
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tove Petersson <tovep@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39214}
2023-01-27 16:32:43 +00:00
Artem Titov
0f2ce5cc1c Revert "Migrate WebRTC documentation to new renderer"
This reverts commit 3eceaf46695518f25bef43f155f82ed174827197.

Reason for revert: 

Original change's description:
> Migrate WebRTC documentation to new renderer
>
> Bug: b/258408932
> Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39205}

Bug: b/258408932
Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39209}
2023-01-26 20:19:12 +00:00
Artem Titov
3eceaf4669 Migrate WebRTC documentation to new renderer
Bug: b/258408932
Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39205}
2023-01-26 14:58:00 +00:00
Philipp Hancke
66efab2dd2 Measure RTCPMuxPolicy at time of connect
to see if we can finally deprecate it.
Chromium metrics CL:
  https://chromium-review.googlesource.com/c/chromium/src/+/4193156

BUG=chromium:713445

Change-Id: I4847620a50f8ece6a2c9aeb5b754b46455e732ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291332
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39203}
2023-01-26 14:06:01 +00:00
Fredrik Hernqvist
5adc2b6969 Correct RTCAudioPlayoutStats type and add kind field.
Bug: webrtc:14653
Change-Id: Idb85ce440620fc5b818a3b23a63ac062a443cc81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291330
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39196}
2023-01-25 14:30:41 +00:00
Philipp Hancke
4893638e72 mark RTPHeader struct RTC_EXPORT
so it can be used in downstream Chrome tests

BUG=None

Change-Id: I4b3e1f172e8eb2ba01ab5c257f3626223781da31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291116
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39163}
2023-01-20 15:46:01 +00:00
Fredrik Hernqvist
828de8036d Populate RTCInboundRtpStreamStats::playoutId when appropriate
Bug: webrtc:14653
Change-Id: I0c59604b218d0839a126c02914626b8ed2bee76c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291040
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39149}
2023-01-19 15:44:36 +00:00
Sergio Garcia Murillo
1389c4b594 Add 444 10 bits support for H264 and VP9
This CL adds support for I410 buffers (444 10 bits) and modify vp9 and h264 for being able to convert input buffer to it when appropiate.

Bug: webrtc:14818
Change-Id: I2fb3dc9d80c5338944c6df74dd6217a0454180d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290721
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39123}
2023-01-17 12:32:26 +00:00
Florent Castelli
a6b9924988 Remove all usage of //rtc_base target
Bug: webrtc:9838
Change-Id: If813dbb426b4dc848185b64c0349d03fa9c059f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290986
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39116}
2023-01-16 14:36:06 +00:00
Fredrik Hernqvist
efbe753617 Add RTCAudioPlayoutStats to GetStats().
This is done by allowing implementations of AudioDeviceModule to
implement the GetStats() method. The default implementation returns
nullopt, in which case RTCAudioPlayoutStats will not be visible in the
stats.

Bug: webrtc:14653
Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39115}
2023-01-16 13:19:45 +00:00
Henrik Boström
3dd73ae6f4 Surface the SetMetadata() method so that Chromium can use it.
RTPVideoHeader is changed to non-const to allow modifying it. We want
to do this when implementing setMetadata() in JavaScript or when
refactoring clone() as "construct + set bytes + setMetadata".

Unblocks
https://chromium-review.googlesource.com/c/chromium/src/+/4164979.

Bug: webrtc:14709
Change-Id: I6089df9c03e9aa33feeb0830dd240dd456cb565e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290981
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39113}
2023-01-16 10:54:17 +00:00
Lionel Koenig
612872b29d Add RtcEvent to store when MinimumSetDelay is set on NetEq
To be able to simulate offline some scenario in which the javascript
layer set the minimum base buffer size of neteq, it is required to
record those API calls. This change introduces this.

Bug: webrtc:14763
Change-Id: Ic817913eda60978d6fca3f8e12229aeec505ca25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287122
Auto-Submit: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39104}
2023-01-13 17:15:48 +00:00
Henrik Boström
dc39aebd08 Add GetRTPVideoHeaderCodecSpecifics() to metadata.
This will allow exposing VP8, VP9 and H264-specific RTP header metadata
in JavaScript (behind a flag).

This information appears to be necessary for cloning
(https://github.com/w3c/webrtc-encoded-transform/issues/161), and
cloning should be the same as "new frame + setMetadata + setBytes",
ergo this should be exposed.

Bug: webrtc:14709
Change-Id: Ie71c05f40689bbd529dc4674a07a87c7910b22d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290880
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39101}
2023-01-13 11:33:40 +00:00
Artem Titov
bb25641dd9 [PCLF] Add an API to add extra audio/video RTP header extensions
Bug: None
Change-Id: Ieee29419bc13efe1891c2ceda8a919c031cd4a58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290897
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39100}
2023-01-13 11:14:38 +00:00
Philipp Hancke
e137c4592e stats: deprecate timestamp_us constructor and method
in favor of the Timestamp constructor and method.
The constructor is most likely not used outside libWebRTC,
the call to
  .timestamp_us()
can be replaced with
  .timestamp().us()

BUG=webrtc:14813

Change-Id: Id166b4f85b2425ecec1c7ebb81406f82ff9d95c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290727
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39066}
2023-01-11 11:40:05 +00:00
Sergio Garcia Murillo
bfc26c65e6 Use libyuv rotate methods
Bug: webrtc:13826
Change-Id: I10a3b291a66eae1b867dd2fa1a1781c235feef33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290703
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39060}
2023-01-10 15:26:37 +00:00
Evan Shrubsole
7ef0c1aff5 Implement RTCNonStandardStatsMember using StatExposureCriteria
Adds a new StatExposureCriteria for non-standard stats. This removes the
virtual call to is_standardized() which can simply use the
StatExposureCriteria.

Bug: webrtc:14546
Change-Id: If4174019ff8cc6559ab0dc9a04e0f8a6631b9842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279045
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39057}
2023-01-10 14:39:39 +00:00
Florent Castelli
a138c6c8a5 Split rtc_base into multiple targets
Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.

Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-09 12:21:25 +00:00