100 Commits

Author SHA1 Message Date
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
Danil Chapovalov
385eb9714d 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}
2022-02-25 16:44:07 +00:00
Stefan Mitic
ffdc6804bf Reland: Added support for H264 YUV444 (I444) decoding.
PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/235340

Bug: chromium:1251096
Change-Id: Icd997c7f7732229954d5494343b4e7a70deb09d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251303
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35964}
2022-02-09 11:57:55 +00:00
Henrik Boström
3f42fdf19f Revert "Added support for H264 YUV444 (I444) decoding."
This reverts commit 3babb8af238a531cbff27951604b09bb78b762cd.

Reason for revert:
- Causes regressions to transceivers, see https://crbug.com/1291956 for more information, including tests to reproduce the issue.

This CL is not a pure revert. While it reverts everything else, it does
keep the new enum value (kProfilePredictiveHigh444). This is as to not
break Chromium which already depend on it. It is not listed in the
kProfilePatterns though so the enum value should never be applicable.

Original change's description:
> Added support for H264 YUV444 (I444) decoding.
>
> Added Nutanix Inc. to the AUTHORS file.
>
> PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/234540
>
> Bug: chromium:1251096
> Change-Id: I99a1b1e4d8b60192ff96f92334a430240875c66c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235340
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35684}

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

Bug: chromium:1251096, chromium:1291956
Change-Id: Ib4d8ea4898f9832914d88e7076e6b39da0c804ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249791
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35835}
2022-01-29 10:45:39 +00:00
Stefan Mitic
3babb8af23 Added support for H264 YUV444 (I444) decoding.
Added Nutanix Inc. to the AUTHORS file.

PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/234540

Bug: chromium:1251096
Change-Id: I99a1b1e4d8b60192ff96f92334a430240875c66c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235340
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35684}
2022-01-13 14:06:55 +00:00
Ilya Nikolaevskiy
54f377308f Revert "Added support for H264 YUV444 (I444) decoding."
This reverts commit 7d8ed3437295bcad945dab69f042cf52a05c0985.

Reason for revert: Breaks internal builds

Original change's description:
> Added support for H264 YUV444 (I444) decoding.
>
> Added Nutanix Inc. to the AUTHORS file.
>
> Bug: chromium:1251096
> Change-Id: Ib47c2b1f94797afb6c5090f3c46eae6f13110992
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234540
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35200}

TBR=ilnik@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,peterhanspers@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com,stefan.mitic@nutanix.com

Change-Id: I3048c353a2b6b4f3d4e5e53a88f48b456f1ce593
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1251096
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235203
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35207}
2021-10-14 12:59:29 +00:00
Stefan Mitic
7d8ed34372 Added support for H264 YUV444 (I444) decoding.
Added Nutanix Inc. to the AUTHORS file.

Bug: chromium:1251096
Change-Id: Ib47c2b1f94797afb6c5090f3c46eae6f13110992
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234540
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35200}
2021-10-14 11:06:55 +00:00
Markus Handell
6fa9e68da9 Route min/max FPS constraints to VideoStreamEncoder.
This change
- adds new type VideoTrackSourceConstraints expressing min/max FPS
  constraints.
- adds new method VideoTrackSourceInterface::ProcessConstraints.
- adds new method VideoSinkInterface<>::OnConstraintsChanged.
- updates AdaptedVideoTrackSource and VideoBroadcaster to forward
  the constraints to sinks.
- adds several unit tests for the added functionality.
- and finally, implements OnConstraintsChanged in VideoStreamEncoder.

Chromium will be updated in coming CLs to supply constraints set
through the MediaStream module.

go/rtc-0hz-present

Bug: chromium:1255737
No-Try: true
Change-Id: Iffef239217269c332a1aaa902ddeae2440929e22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235040
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35197}
2021-10-14 06:49:05 +00:00
philipel
10dc1a6d8b New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class.
Bug: webrtc:12579
Change-Id: Idea35983e204e4a3f8628d5b4eb587bbdbff5877
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227286
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34999}
2021-09-15 09:57:29 +00:00
Danil Chapovalov
ecc46eff5b Introduce new api to initialize VideoDecoder
Bug: webrtc:13045
Change-Id: If14fa3998176ee07b6f2835745568f70347ccac6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227766
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34694}
2021-08-10 08:42:43 +00:00
Byoungchan Lee
f740c252e7 Use the underlying type of Java Video Buffer on Java -> C++ Frame Buffer
Just like the C++ API, add a method in Java VideoFrame.Buffer that
describes the underlying implementation.
Use this method to properly select AndroidVideoBuffer
or AndroidVideoI420Buffer in Java -> C++ Video Frame Conversion.

Also, add a test case for WrappedNativeI420Buffer
in VideoFrameBufferTest for consistency.

Bug: webrtc:12602
Change-Id: I4c0444e8af6f6a1109bc514e7ab6c2214f1f6d60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223080
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34545}
2021-07-24 01:04:40 +00:00
Byoungchan Lee
9fc2663712 Hide VideoCodecType from Android SDK
This has not been used since
https://webrtc-review.googlesource.com/c/src/+/172721 .

Bug: None
Change-Id: Id617b9f6770b342b324fe0da84bf402cea1e783c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223081
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/master@{#34480}
2021-07-15 18:33:47 +00:00
philipel
e9a74c918b Public RtpVideoFrameAssembler
This class takes RtpPacketReceived and assembles them into RtpFrameObjects.

Change-Id: Ia9785d069fecccc1d5b81efd257f33c8bd7a778b
Bug: webrtc:7408, webrtc:12579
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222580
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34364}
2021-06-24 15:20:42 +00:00
Henrik Boström
f412976eca Provide a default implementation of NV12BufferInterface::CropAndScale.
This avoids falling back on the VideoFrameBuffer::CropAndScale default
implementation which performs ToI420. This has two major benefits:
1. We save CPU by not converting to I420 for NV12 frames.
2. We make is possible for simulcast encoders to use Scale() and be
   able to trust that the scaled simulcast layers have the same pixel
   format as the top layer, which is required by libvpx.

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

Bug: webrtc:12595, webrtc:12469
Change-Id: I81aac5c6b3e81c49f32a7be6dc2640e6b40f7692
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212643
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33521}
2021-03-22 11:09:36 +00:00
Danil Chapovalov
e904161cec Replace RTC_DEPRECATED with ABSL_DEPRECATED
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.

Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
2021-02-22 12:53:23 +00:00
Per Kjellander
b03b6c8a94 Move setting of encoder bitrate allocation callback type to VideoSendStream
It turned out that the negotiated rtp header extensions are not fully known in WebRtcVideoChannel::AddSendStream.

The cl also remove the unnecessary factory for creating VideoStreamEncoder.


Bug: webrtc:12000
Change-Id: If994c8deb69f3ce4212896d3ad757dac94c6e09f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198840
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32916}
2021-01-07 09:29:05 +00:00
Danil Chapovalov
9c99b7964f Use SvcRateAllocator for av1
same as VP9, Av1 encoder supports spatial scalability and thus
SvcRateAllocator better fits for it than SimulcastRateAllocator

Bug: webrtc:12148
Change-Id: I3f78afb3aec00b6a8a7242fe8dce07752e7a514e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191960
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32565}
2020-11-06 10:23:17 +00:00
Per Kjellander
a94348440b VideoStreamEncoder report VideoLayersAllocation for simulcast
Adds support for Vp8 simulcast.

Bug: webrtc:12000
Change-Id: Ib24fd0542642b023ec35f7a7bdc4880d72365edf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187341
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32416}
2020-10-15 17:39:06 +00:00
Ilya Nikolaevskiy
113326bbe9 Reland "Remove deprecated video_frame_i420 build target"
No changes after the fix in the downstream project.

Bug: chromium:1132299
Change-Id: I7cb490d9e8ead7aea758f9542a84268fd4160016
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187497
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32390}
2020-10-13 13:00:12 +00:00
Ilya Nikolaevskiy
91dddf3707 Revert "Remove deprecated video_frame_i420 build target"
This reverts commit 33bd4fbe1e5b0ddf350409431ee5d9bb4e2a8132.

Reason for revert: Breaks downstream project

Original change's description:
> Remove deprecated video_frame_i420 build target
>
> Bug: chromium:1132299
> Change-Id: Ie0207037b46d4c40d2d808bf5fcddf71f43e325b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187350
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32385}

TBR=mbonadei@webrtc.org,ilnik@webrtc.org,mbonadei@chromium.org

Change-Id: I83185a905191877c5b8e9826cc3555fa9406a537
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1132299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187496
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32386}
2020-10-13 09:01:04 +00:00
Ilya Nikolaevskiy
33bd4fbe1e Remove deprecated video_frame_i420 build target
Bug: chromium:1132299
Change-Id: Ie0207037b46d4c40d2d808bf5fcddf71f43e325b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187350
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32385}
2020-10-13 08:30:52 +00:00
Ilya Nikolaevskiy
38e9b06151 Reland "Add scaling interface to VideoFrameBuffer"
(Reland with no changes after the fix to the downstream project)

This can be overriden for kNative frame types to perform scaling efficiently.

Default implementations for existing buffer types require actual
buffer implementation, thus this CL also merges "video_frame"
with "video_frame_I420" build targets.

Originally Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303

(Landing with TBR as it's unchaged reland of already approved CL)
TBR=nisse@webrtc.org,sakal@webrtc.org

Bug: webrtc:11976, chromium:1132299
Change-Id: Ia23f7d3e474bd9cdc177104cc5c6d772f04b210f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187345
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32362}
2020-10-09 08:30:50 +00:00
Ilya Nikolaevskiy
441dbf9a56 Revert "Add scaling interface to VideoFrameBuffer"
This reverts commit c79f1d8cfb48c348692320b05fa31598ed7067ca.

Reason for revert: Breaks downstream project.

Original change's description:
> Add scaling interface to VideoFrameBuffer
>
> This can be overriden for kNative frame types to perform scaling efficiently.
>
> Default implementations for existing buffer types require actual
> buffer implementation, thus this CL also merges "video_frame"
> with "video_frame_I420" build targets.
>
> Bug: webrtc:11976, chromium:1132299
> Change-Id: I3bf5f6bf179db5e7ab165b1c2301980043a08765
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#32352}

TBR=mbonadei@webrtc.org,sakal@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,eshr@google.com

Change-Id: I86ac697bf963ef7e2c4f2ed34c3a7bf04f4f1ce1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11976
Bug: chromium:1132299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187344
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32354}
2020-10-08 14:16:23 +00:00
Ilya Nikolaevskiy
c79f1d8cfb Add scaling interface to VideoFrameBuffer
This can be overriden for kNative frame types to perform scaling efficiently.

Default implementations for existing buffer types require actual
buffer implementation, thus this CL also merges "video_frame"
with "video_frame_I420" build targets.

Bug: webrtc:11976, chromium:1132299
Change-Id: I3bf5f6bf179db5e7ab165b1c2301980043a08765
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32352}
2020-10-08 13:33:00 +00:00
Per Kjellander
6556ed2402 Add experimental extension RtpVideoLayersAllocation
The extension is suggested to be used for signaling per target bitrate, resolution
and frame rate to a SFU to allow a SFU to know what video layers a client is currently targeting.
It is hoped to replace the current Target bitrate RTCP XR message currently used only for screen share.

Bug: webrtc:12000
Change-Id: Id7b55e7ddaf6304e31839fd0482b096e1dbe8925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185980
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32313}
2020-10-05 13:38:13 +00:00
philipel
3dc4780d8e Added VideoContentType to OnDecodedFrame callback.
Also added a FrameInfo struct to propagate various meta information along side it.

Bug: webrtc:9106
Change-Id: I1feb9f94c662c367f7c6e0a50d33705fdd5346bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183880
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32079}
2020-09-11 10:22:05 +00:00
Evan Shrubsole
84995439fd Add NV12 video buffer type
This will allow incoming NV12 frames to be encodable by libvpx without
requiring a conversion to I420 before encoding. NV12 is supported in
libvpx https://chromium.googlesource.com/webm/libvpx/+/master/CHANGELOG

Bug: webrtc:11916
Change-Id: I30e9c42c0607bee07691930c0248921bba09134c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183720
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32061}
2020-09-09 15:26:02 +00:00
Niels Möller
d381eede92 Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
We can then finally delete the top-level common_types.h, and the
corresponding build target webrtc_common.

Bug: webrtc:7660
Change-Id: I1c1096541477586d90774c7a3405b9d36edec14a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182800
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32044}
2020-09-07 08:37:14 +00:00
Niels Möller
cccd55094d Delete unneeded dependencies on deprecated build target webrtc_common
Bug: webrtc:7660
Change-Id: Iad32aad8432fa2c6b3018d511b51943f869fbd11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182420
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31986}
2020-08-25 07:33:12 +00:00
philipel
9465978a3b Remove framemarking RTP extension.
BUG=webrtc:11637

Change-Id: I47f8e22473429c9762956444e27cfbafb201b208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176442
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31522}
2020-06-15 11:18:00 +00:00
Henrik Boström
f4a9991cce [Adaptation] Adding adaptation resources from Call.
This CL adds AddAdaptationResource to Call and
AddAdaptationResource/GetAdaptationResources method to relevant
VideoSendStream and VideoStreamEncoder interfaces and implementations.

Unittests are added to ensure that resources can be added to the Call
both before and after the creation of a VideoSendStream and that the
resources always gets added to the streams.

In a follow-up CL, we will continue to plumb the resources all the way
to PeerConnectionInterface, and an integration test will then be added
to ensure that injected resources are capable of triggering adaptation.

Bug: webrtc:11525
Change-Id: I499e9c23c3e359df943414d420b2e0ce2e9b2d56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177002
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31499}
2020-06-11 12:43:21 +00:00
Mirko Bonadei
2dcf348011 Use absl_deps in order to preapre to the Abseil component build release.
Bug: webrtc:1046390
Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-08 12:59:40 +00:00
Henrik Boström
f0eef12e68 [Adaptation] Add more ResourceAdaptationProcessor logging.
This should help debugging when adaptation is or is not happening
unexpectedly. Log spam is prevented by not logging if the same
result happened to the same resource already and we haven't
adapted since then.

Bug: webrtc:11616
Change-Id: Ia6c5cc35061d252f1c66f2f2bf3b94d2485498d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176221
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31378}
2020-05-28 15:06:46 +00:00
Marina Ciocea
cdc89b4d14 Add GetMetadata() to TransformableVideoFrameInterface API.
Define VideoHeaderMetadata, containing a subset of the metadata in RTP
video header, and expose it the TransformableVideoFrameInterface, to
enable web application to compute additional data according to their own
logic, and eventually remove GetAdditionalData() from the interface.

Bug: chromium:1069295
Change-Id: Id85b494a72cfd8bdd4c0614844b9f0ffae98c956
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174822
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31265}
2020-05-14 19:26:55 +00:00
Evan Shrubsole
ce0a11d5f9 Unify AdaptationReason and AdaptReason enums.
Moves the unified AdaptationReason to the api/ folder.

Bug: webrtc:11392
Change-Id: I28782e82ef6cc3ca3b061f65b0bbdc3766df1f9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172583
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31084}
2020-04-16 13:33:49 +00:00
Evan Shrubsole
dff792591f Remove VideoStreamEncoderObserver::AdaptationReason::kNone
Replaces this with 2 methods instead, adding clarity.

ClearAdaptationStats
- Resets the adaptations statistics to 0. This is done,
when the degredation is reset, for example when the preference
is changed to/from BALANCED.

UpdateAdaptationMaskingSettings
- Updates the settings for adaptation statistics reporting.
This way we don't report quality adaptations if quality scaling
is not enabled (same for resolution/fps scaling).

The adaptation counting inside the SendStatisticsProxy is
now done in a struct that counts the totals, and then masks
out these counts based on the adaptation settings. The
MaskedAdaptationSteps uses optionals to hide the values we
shoudn't report, while the AdaptationSteps always hold the real
totals.

All tests have been updated to use the Reset/Clear method as needed.

Now that AdaptationCounters and AdaptSteps use the same structure,
AdaptationCounters was moved to api/video and replaces AdaptSteps.

The AdaptReason enum is also redundant now, and will be removed
in a follow-up CL.

R=hbos@webrtc.org

Bug: webrtc:11392
Change-Id: Iaed6488581325d341a056b5bbf76a01c19d6c282
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171685
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31083}
2020-04-16 13:27:50 +00:00
Evan Shrubsole
c70b1028d4 Move AdaptationCounters from video/ to api/
- Rename AdaptationCounters to VideoAdaptationCounters
- Move VideoAdaptationCounters to the api/ folder
- Move related tests to api/test/ folder
- Remove VideoAdaptationCounters::operator-

Bug: webrtc:11392
Change-Id: I0de2537e9c8dd9cf29a2ecceee00f92a5b155c83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172920
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31006}
2020-04-06 13:27:28 +00:00
Henrik Boström
62057627ef [Adaptation] Rename and move resource adaptation module/processor stuff.
Rename:
- call/adaptation/resource_adaptation_module_interface.[h/cc] -->
  call/adaptation/resource_adaptation_processor_interface.[h/cc]
- call/adaptation/resource_adaptation_processor.[h/cc] -->
  call/adaptation/new_resource_adaptation_processor_poc.[h/cc]

Move + Rename:
- video/overuse_frame_detector_resource_adaptation_module.[h/cc] -->
  video/adaptation/resource_adaptation_processor.[h/cc]

Move:
- video/encode_usage_resource.[h/cc] --> video/adaptation/...
- video/overuse_frame_detector.[h/cc] --> video/adaptation/...
- video/quality_scaler_resource.[h/cc] --> video/adaptation/...

Unittests are also moved. In order to avoid a circular dependency,
VideoStreamEncoder::kDefaultLastFrameInfo[Width/Height] is moved and
renamed to kDefaultInputPixels[Width/Height] in
video/adaptation/resource_adaptation_processor.[h/cc].

Bug: webrtc:11222
Change-Id: Icf920e8a7362002b1c63c42b2d9e2e63c990b532
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170117
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30754}
2020-03-11 09:30:16 +00:00
Danil Chapovalov
59f3b71c04 Automate conversion from c++ VideoCodeType to java VideoCodecType
Bug: b/148146536
Change-Id: I030c7c6c2a1a9d002bcc60f45c8d6025bd0935b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167301
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30751}
2020-03-11 08:02:36 +00:00
Danil Chapovalov
64f1f3f04e Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED
Bug: None
Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 15:20:35 +00:00
Markus Handell
026f64fc54 Revert "VideoFrame: Store a reference to an encoded frame"
This reverts commit e6eded31e642b3b986fef478315603b5f398c227.

Reason for revert: A better method for communicating encoded frames in VideoTrackSourceInterface surfaced.

Original change's description:
> VideoFrame: Store a reference to an encoded frame
> 
> Enable webrtc::VideoFrame to store a reference to an encoded frame.
> 
> Bug: chromium:1013590
> Change-Id: Id5a06f1c7249f104dfd328f08677cf8001958f0d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158788
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> 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/master@{#29809}

TBR=ilnik@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,handellm@webrtc.org

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

Bug: chromium:1013590
Change-Id: I46384b7997e7b1cd3a2a2042cf17890fc977cca3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160204
Reviewed-by: Markus Handell <handellm@webrtc.org>
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/master@{#29863}
2019-11-21 14:55:21 +00:00
Markus Handell
9982efa830 MediaStreamInterface: introduce encoded sinks.
This change adds a new type of sink for consuming encoded data from
a video source.

Bug: chromium:1013590
Change-Id: Ia7c4e372190c3d6bc007a0d4deb05c2d1bce58d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159927
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29856}
2019-11-21 12:03:35 +00:00
Markus Handell
e6eded31e6 VideoFrame: Store a reference to an encoded frame
Enable webrtc::VideoFrame to store a reference to an encoded frame.

Bug: chromium:1013590
Change-Id: Id5a06f1c7249f104dfd328f08677cf8001958f0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158788
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
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/master@{#29809}
2019-11-16 10:15:38 +00:00
Mirko Bonadei
86d053c2db Use source_sets in component builds and static_library in release builds.
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).

Source sets always pass all the object files to the linker.

On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.

See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set

Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:17:18 +00:00
Mirko Bonadei
35214fcfe2 Add missing RTC_EXPORT for the component build.
Bug: webrtc:9419
Change-Id: I3225259fb4cc55e9820f590928795f4587f1e3cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29479}
2019-10-15 09:07:44 +00:00
Niels Möller
b9bfe655d4 Delete VCMEncodedFrame::VerifyAndAllocate
And mark EncodedImage::Allocate as deprecated.

Bug: webrtc:9378
Change-Id: I03ce907fa6b87803ddb72f548f60a9bf1b7c317d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155163
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29383}
2019-10-04 14:16:49 +00:00