145 Commits

Author SHA1 Message Date
Tommi
f1656def75 Apply dimension checking across the YUV buffer classes.
Bug: chromium:371686447
Change-Id: I15502748c0b0036aaef3742bce27104887b77f65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372140
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43617}
2024-12-19 08:31:15 -08:00
Tommi
0dec0897f1 Make I420DataSize trigger a crash in case of int overflow.
Bug: chromium:371686447
Fixes: chromium:371686447
Change-Id: Icd4ef5f1edc54853445bb1542eff62e354655368
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43588}
2024-12-17 07:25:34 -08:00
Erik Språng
e5f6f1fab4 Add optional corruption filter settings to EncodedImage.
This is a prerequisite for enabling implementation-specific filter
settings for automatic corruption detection.

Bug: webrtc:358039777
Change-Id: I363c592aa35164f690dd4ad1204e90afc0277d8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368940
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43443}
2024-11-22 12:10:31 +00:00
Qiu Jianlin
ff9e7cb182 Include H.265 support in RTP video frame assembler.
This adds support of H.265 into the RTP video frame assembler, which is
now a public interface.

Bug: chromium:41480904
Change-Id: I74fd761949d0b095ba4526d2fa887e963f48abcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367603
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43374}
2024-11-08 00:38:38 +00:00
Sergio Garcia Murillo
e17aad2c1d Use rtc::Buffer for memory storage of EncodedImageBuffer
The goal is to be able to write the rtc::Buffer by another utility
(like rtc::ByteBufferWriter) and pass it into EncodedImageBuffer
without memcpy.

Bug: webrtc:42223344
Change-Id: Ieda55e77a36636e8cdff6ad6b7d078de0aeafec0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364243
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43179}
2024-10-07 11:06:04 +00:00
Florent Castelli
8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00
Dor Hen
9fb83a18e3 Apply include-cleaner to api/video
Bug: webrtc:42226242
Change-Id: I023f058f3b5e2747bd02f01a191a91636c85f12d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42801}
2024-08-19 11:22:41 +00:00
Florent Castelli
99c519b3fd Mass removal of absl_deps in all BUILD.gn files
Bug: webrtc:341803749
Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 15:09:46 +00:00
Danil Chapovalov
fd89ff5d93 Provide Environment to SimulcastRateAllocator at construction
So that this class can use propagated field trials instead of the global

Bug: webrtc:42220378
Change-Id: Ic1dba0c4967735606904329f7e9e6c09f186b809
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350641
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42326}
2024-05-16 13:32:54 +00:00
Danil Chapovalov
179f96de7e Provide Environment to construct VideoBitrateAllocator
To allow various VideoBitrateAllocators to use propagated rather than global field trials

This relands the
https://webrtc-review.googlesource.com/c/src/+/349920
where patchset#1 is identical to the original change,
patchset#2 undoes (postpones) the expectation downstream propagates the Environment too.

Bug: webrtc:42220378
Change-Id: I4a9a32bb0926a875d37f3ba19dd5309e97546553
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350364
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42298}
2024-05-14 11:36:42 +00:00
Jeremy Leconte
16fb7903e5 Revert "Provide Environment to construct VideoBitrateAllocator"
This reverts commit 4bf4e1753ce1567aab9f83f3a324e8bda2833459.

Reason for revert: break upstream 

Original change's description:
> Provide Environment to construct VideoBitrateAllocator
>
> To allow various VideoBitrateAllocators to use propagated rather than global field trials
>
> Bug: webrtc:42220378
> Change-Id: I52816628169a54b18a4405d84fee69b101f92f72
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349920
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42288}

Bug: webrtc:42220378
Change-Id: I7d47eb635c2d312d97a870c2a8eca0b23d2f86a0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350307
Owners-Override: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42290}
2024-05-13 13:32:28 +00:00
Danil Chapovalov
4bf4e1753c Provide Environment to construct VideoBitrateAllocator
To allow various VideoBitrateAllocators to use propagated rather than global field trials

Bug: webrtc:42220378
Change-Id: I52816628169a54b18a4405d84fee69b101f92f72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349920
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42288}
2024-05-13 12:12:01 +00:00
Qiu Jianlin
9c95a4f704 Helper API for codec factories to calculate supported H.265 levels.
This expose a new GetSupportedH265Level API for WebRTC external
factories to calculate H.265 levels to be use for SDP negotation.

Bug: webrtc:13485
Change-Id: Ib420da2b9b1b7af00129294be5b3efec172e8faf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345544
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42079}
2024-04-16 09:23:58 +00:00
Harald Alvestrand
78f905e5cc Move some users to use webrtc::RefCountInterface
Bug: webrtc:15622
Change-Id: I2d4c20c726af1a052e161b7689a73d1e5e3eb191
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325526
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41067}
2023-11-02 14:45:57 +00:00
Tony Herre
5f14f9e6ed Remove VCMEncodedFrame from webrtc::EncodedFrame inheritance
Remove VCMEncodedFrame from the inheritance chain of EncodedFrames by
- moving getters for EncodedImage fields up to EncodedImage
- copying other non-deprecated fields & Methods from VCMEncodedFrame over to EncodedFrame
- Removing EncodedFrame's inheritance of VCMEncodedFrame

We leave VCMEncodedFrame as part of the (near) deprecated
VideoCodingModule code. The only place which needs to accept either is
in the generic decoder.

Bug: webrtc:9378, b:296992877
Change-Id: I60706aebbb6eacc7fd4b35ec90cc903efdbe14c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317160
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40639}
2023-08-28 11:46:48 +00:00
Harald Alvestrand
00f11224fd Remove extra usage of video-content-type header extension
This extension is documented to carry one bit: Screenshare.
It's been used for carrying simulcast layers and experiment IDs.
This CL removes that usage.

Bug: webrtc:15383
Change-Id: I048b283cde59bf1f607d8abdd53ced07a7add6f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40457}
2023-07-22 21:47:08 +00:00
Danil Chapovalov
630c40d716 Update RtpSenderVideo::SendVideo/SendEncodedImage to take Timestamp/TimeDelta types
Bug: webrtc:13757
Change-Id: I2f21b14ecf003c5cb0c4c92d0c6b9b6f11c35f71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311945
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40450}
2023-07-21 10:36:49 +00:00
Jianhui Dai
32a8169a65 Use common VideoFrameTypeToString helper
This CL cleans up all local conversions, in favor of the common helper
function.

Bug: webrtc:15210
Change-Id: Id77e1c6b1151a2542d92e220e91d5b11285479b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311060
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40420}
2023-07-12 00:28:47 +00:00
Rasmus Brandt
cde5354729 Implement DelayVariationCalculator for events analysis.
This CL implements {,Logging}DelayVariationCalculator, whose purpose is to calculate simple inter-arrival metrics for a sequence of RTP frames. Uses could include RtcEventLog analysis and ad hoc testing.

Want lgtm: asapersson

Bug: webrtc:15213
Change-Id: I3f9d13a2c4fa66b6f1229c1b6fcd66a6911070de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306741
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40247}
2023-06-08 17:42:53 +00:00
Tove Petersson
1fccaa4485 Reland "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
This reverts commit 8bf321062973939ef35f529640f5e69852e89a7e.

Reason for revert: Initialized an uninitialized member in GofInfoVP9 (+ removed some redundant initialization of members already initialized by SetGofInfoVP9())

Original change's description:
> Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
>
> This reverts commit 437bf78ed9518b21fc39b94f6ee42d5b157e6084.
>
> Reason for revert: Breaks upstream project
>
> Original change's description:
> > operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
> >
> > Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
> >
> > Also default-initialized VideoFrameMetadata::ssrc_ to 0.
> >
> > Bug: webrtc:14708
> > Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
> > Commit-Queue: Tove Petersson <tovep@google.com>
> > Reviewed-by: Tony Herre <herre@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39411}
>
> Bug: webrtc:14708
> Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39413}

Bug: webrtc:14708
Change-Id: I843d29f7dd0da2c7f16968a7fc08dc02cd359fc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tove Petersson <tovep@google.com>
Cr-Commit-Position: refs/heads/main@{#39418}
2023-02-28 15:44:21 +00:00
Andrey Logvin
8bf3210629 Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
This reverts commit 437bf78ed9518b21fc39b94f6ee42d5b157e6084.

Reason for revert: Breaks upstream project

Original change's description:
> operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
>
> Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
>
> Also default-initialized VideoFrameMetadata::ssrc_ to 0.
>
> Bug: webrtc:14708
> Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
> Commit-Queue: Tove Petersson <tovep@google.com>
> Reviewed-by: Tony Herre <herre@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39411}

Bug: webrtc:14708
Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39413}
2023-02-28 11:50:42 +00:00
Tove Petersson
437bf78ed9 operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.

Also default-initialized VideoFrameMetadata::ssrc_ to 0.

Bug: webrtc:14708
Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
Commit-Queue: Tove Petersson <tovep@google.com>
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39411}
2023-02-28 08:32:09 +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
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
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
Evan Shrubsole
097fc347ec [Unwrap] Prepare SequenceNumberUnwrapper for migrations
This is in prep for the migration of all unwrappers to
SequenceNumberUnwrapper as a standard implementation.

This moves the SeqNumUnwapper to its own header and adds 2 methods to
SeqNumUnwrapper which are defined by other unwrappers:
* PeekUnwrap
* Reset

It also adds two implementations for RtpTimestamps and
RtpSequenceNumbers.

Bug: webrtc:13982
Change-Id: I5baefb2de1db92fe1bb600760bd63b71e9310eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288742
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39030}
2023-01-09 11:42:20 +00:00
Henrik Boström
13730e9742 Rename VideoFrameMetadata tests to RTPVideoHeaderTest.
This is a pure move/rename. The reason for wanting the tests in
RTPVideoHeader is that it is the GetAsMetadata() function that we are
testing and in a future CL we'll also want to test SetFromMetadata().

// Bots green, no need to wait for the remaining ones, just a move
NOTRY=True

Bug: webrtc:14709
Change-Id: Iecb938e79e7e8d55e208baea190eef4c6730158e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285460
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38764}
2022-11-29 16:03:20 +00:00
Henrik Boström
bf2f605e03 Add more information to RTPVideoHeader::GetAsMetadata().
Update GetAsMetadata() to include more of the RTPVideoHeader metadata.
The intent is to be able to both get and set all of these from
JavaScript behind a flag.

Planned follow-up CLs:
1. Also get codecs-specifics, starting with VP8.
2. Test refactoring/rename: Move tests to RTPVideoHeaderTest.
3. Add RTPVideoHeader::SetFromMetadata() covering everything gettable.
4. Chrome plumbing.

Bug: webrtc:14709
Change-Id: I78679b9ff4ca749d50f309a1713e71ceabb826dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285084
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38756}
2022-11-29 12:30:46 +00:00
Henrik Boström
158d5e3078 Add RTPVideoHeader::GetAsMetadata().
In preparation of adding RTPVideoHeader::SetFromMetadata() method, the
VideoFrameMetadata construct-from-RTPVideoHeader is replaced by
RTPVideoHeader::GetAsMetadata(). This serves two purposes:
1. Having "GetAs" and "SetFrom" in the same file reduces the risk of
   these two methods getting out of sync as we expand its usage.
2. This is necessary to avoid a circular dependency that would
   otherwise be introduced by RTPVideoHeader::SetFromMetadata().

Bug: webrtc:14709
Change-Id: I127b3d15f9a8c6af210449a5a50d414c9ba79930
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285080
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38735}
2022-11-25 14:40:30 +00:00
Evan Shrubsole
20afff9263 Expose frame_buffer GN target
Bug: None
Change-Id: I75068b87e95575235eb937ef73279f961d0df93e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282322
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38571}
2022-11-07 17:32:57 +00:00
Jonas Oreland
6c2dae21e9 Move VideoEncoderConfig from api/ into video/config
This cl move VideoEncoderConfig from api/ to video/config.

VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.

brandt@ think that the reason these were in api/ in the
first place had to downstream project.

Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).

Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
2022-09-29 09:44:43 +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
Sergio Garcia Murillo
8545ebae28 Add 420 and 422 10 bit h264 decoding.
422 10 bit format is called I210 in the code and implemented in I210Buffer, and 420 10-bit format format is using is using the already existing I010 format and implemented in I010Buffer.

Bug: webrtc:13826
Change-Id: I6b6ed65b9fbb295386ea20f751bd0badc49ef21b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256964
Reviewed-by: Niels Moller <nisse@webrtc.org>
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@{#37252}
2022-06-17 11:12:10 +00:00
Niels Möller
7c8c4db8ea Add rtc::make_ref_counted to api/
This cl adds a forwarding header, a build target, and migrates headers
in api/ to use it.

Moving actual implementation, will follow, in
https://webrtc-review.googlesource.com/c/src/+/265390.

Bug: webrtc:12701
Change-Id: Id950725d7d054de8a83b3800b9c9a6437344de86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37200}
2022-06-13 15:53:27 +00:00
philipel
8615bf0582 Move FrameBuffer3 to api/
The webrtc::VideoStreamDecoderInterface was basically created as a public version of FrameBuffer2, but to hide the complexity of FrameBuffer2 it was also combined with decoding so that the public API could be reasonably simple to use. FrameBuffer3 has a simple API with a clear purpose, so its API can be exposed directly.

Bug: webrtc:14026
Change-Id: I81dc84b869e4d16c5e02feb5c876fbcede3d4a25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261181
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36781}
2022-05-05 14:34:48 +00:00
Florent Castelli
c3e6e3a3e8 Remove dependency on rtc_base_approved from most targets
Bug: webrtc:9838
Change-Id: Ibd0199803597eff48ca139a5cecdc3209c62c5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259873
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36643}
2022-04-25 12:15:30 +00:00
Florent Castelli
dd837e28fa Remove //rtc_base:timeutils from public deps
Bug: webrtc:8603
Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-06 11:23:21 +00:00
Florent Castelli
57aa81bce7 Remove //rtc_base:stringutils from public deps
Bug: webrtc:8603
Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-05 22:42:19 +00:00
Florent Castelli
e10a9f609a Remove //rtc_base:safe_conversions from public deps
Bug: webrtc:8603
Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-05 20:04:59 +00:00
Florent Castelli
f86f6f9afd Remove //rtc_base:refcount from public deps
Bug: webrtc:8603
Change-Id: Ib27a107ae809df739492846175f0e9c4af40d21a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257910
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36447}
2022-04-05 15:32:29 +00:00
Florent Castelli
0af55ba60d Remove //rtc_base:logging from public deps
Bug: webrtc:8603
Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-05 10:31:19 +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
Sergio Garcia Murillo
b63536f5d3 add h264 422 decoding
Bug: webrtc:13826
Change-Id: Ic7296be69157a9aaf5f139a18fdb011b90f4caa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255380
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36337}
2022-03-25 13:15:34 +00:00
Jonas Oreland
e02f9eedb3 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 10/inf
This patch takes a stab at modules/video_coding,
but reaches only about half.

Bug: webrtc:10335
Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36335}
2022-03-25 12:35:36 +00:00
Evan Shrubsole
d6cdf80072 Use Timestamp and TimeDelta in VCMTiming
* Switches TimestampExtrapolator to use Timestamp as well.

Bug: webrtc:13589
Change-Id: I042be5d693068553d2e8eb92fa532092d77bd7ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249993
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36112}
2022-03-02 15:07:25 +00:00
Danil Chapovalov
9af4aa7cf4 Reland "Represent RtpPacketToSend::capture_time with Timestamp"
This reverts commit 56db8d09529d5ba92d24954a1d78a90c8ea2cd85.

Reason for revert: downstream problem addressed

Original change's description:
> Revert "Represent RtpPacketToSend::capture_time with Timestamp"
>
> This reverts commit 385eb9714daa80306d2f92d36678c42892dab555.
>
> Reason for revert: Causes problems downstream:
>
> #
> # Fatal error in: rtc_base/units/unit_base.h, line 122
> # last system error: 0
> # Check failed: value >= 0 (-234 vs. 0)
>
> Original change's description:
> > Represent RtpPacketToSend::capture_time with Timestamp
> >
> > Bug: webrtc:13757
> > Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36083}
>
> Bug: webrtc:13757
> Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36087}

Bug: webrtc:13757
Change-Id: I1fa852757480116f35deb2b6c3c27800bdf5e197
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252781
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36093}
2022-02-28 10:04:37 +00:00
Tomas Gunnarsson
56db8d0952 Revert "Represent RtpPacketToSend::capture_time with Timestamp"
This reverts commit 385eb9714daa80306d2f92d36678c42892dab555.

Reason for revert: Causes problems downstream:

#
# Fatal error in: rtc_base/units/unit_base.h, line 122
# last system error: 0
# Check failed: value >= 0 (-234 vs. 0)

Original change's description:
> Represent RtpPacketToSend::capture_time with Timestamp
>
> Bug: webrtc:13757
> Change-Id: I0ede22cd34e3a59afe1477d8edd495dce64e3242
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252586
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36083}

Bug: webrtc:13757
Change-Id: I8442abd438be8726cf671d0f372d50ecfac6847e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252720
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36087}
2022-02-26 10:35:13 +00:00