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}
Additionally,
* Moved to its own GN target.
* Added unittests.
* Removed unused variable `_zeroWallClock`.
* Renamed variables to match style guide.
* Moved fields _dTS and _wrapArounds to variables.
Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d
Bug: webrtc:13756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36147}
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}
This reverts commit 3babb8af23.
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}
Currently if encoder initialization fails WebRTC doesn't send any video.
This CL adds functionality that changes encoder type in such case and
restores the video. If encoder selector is available we switch to
encoder it recommends. Otherwise, VP8 is used as the default fallback
encoder.
Bug: webrtc:13572
Change-Id: Ifcdf707a575711f5ff81f9451caf30140c9171dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246960
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35761}
Updates all webrtc code, to have a small followup cl to just add the
"explicit" keyword. Patchset #24 passed all webrtc tests, with explicit.
Bug: webrtc:13464
Change-Id: I39863d3752f73209b531120f66916dc9177bf63a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242363
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35718}
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}
Under zero-hertz mode, provided that a frame arrived to the
VideoStreamEncoder, the receiver may experience up to a second
between incoming frames. This results in key frame requests getting
serviced with that delay, which is undesired.
What's worse is also the fact that if no frame ever arrived to the
VideoStreamEncoder, it will not service the keyframe requests at all
until the first frame comes.
This change introduces VideoSourceInterface::RequestRefreshFrame
which results in a refresh frame being sent from complying sources.
The method is used under zero-hertz mode from the VideoStreamEncoder
when frames didn't arrive to it yet (with changes to the zero-hertz
adapter).
With this change, when the frame adapter has received at least one
frame, it will conditionally repeat the last frame in response to the
key frame request.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I6f97813b3a938747357d45e5dda54f759129b44d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242361
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35562}
The frame cadence adapter previously resulted in unconditional
frame repeating at max FPS. Change this to slow down to an idle
rate (1 Hz) when quality convergence in all configured spatial
layers has been achieved.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: Ifa593dbf8a61aa29da20ac250da332734ae82791
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241421
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35547}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
This change introduces a new FrameCadenceAdapter class which takes the
role of being a VideoFrameSinkInterface<> instead of VideoStreamEncoder.
The FrameCadenceAdapter will see its functionality grow in future CLs
and eventually enable screenshare capture sources to have zero hertz as
the minimum capture frequency.
This CL moves logic related to UMA collection and constraints into the
adapter.
The adapter has two major modes. Future functionality is planned to be
added under the WebRTC-ZeroHertzScreenshare field trial. Unit tests are
added that verify passthrough operation when WebRTC-ZeroHertzScreenshare
isn't specified or disabled.
Just specifying the WebRTC-ZeroHertzScreenshare field trial isn't
enough to activate the feature, but the caller has to additionally
configure screen content type, minimum FPS 0, and maximum FPS > 0 for
the new mode.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I1799110ed40843152786ad80df10acfb83a608b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236682
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35315}
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}
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}
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}
Bilinear is faster but lesser quality, box is best quality. Our code
base has disagreed about which filter to use for quite some time,
causing aliasing bug reports. In an effort to avoid aliasing artifacts
and make our scaling filters more predictable, we're updating all uses
to kFilterBox.
WebRTC already uses kFilterBox everywhere except for these three
places. The main discrepency was between Chromium and WebRTC but that
has already been fixed. This CL fixes the last remaining bilinears.
This brings the WebRTC kFilterBox use count up from 11 to 14 and the
kFilterBilinear use count down from 3 to 0.
Bug: chromium:1212630
Change-Id: I5fe4aa92b9275d65b91ea97925533055d190d317
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221372
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34248}
This change adds support for emitting encoded frames
for recording when the decoder can't easily read out
encoded width and height as is the case for AV1 streams,
in which case the information is buried in OBUs. Downstream
project relies on resolution information being present for key
frames. With the change, VideoReceiveStream2 infers the
resolution from decoded frames, and supplies it in the
RecordableEncodedFrames.
Bug: chromium:1191972
Change-Id: I07beda6526206c80a732976e8e19d3581489b8fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214126
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33662}
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}
This will allow us to optimize the internal buffers of
webrtc::VideoFrame for the resolution(s) that we actually want to
encode.
Bug: webrtc:12469, chromium:1157072
Change-Id: If378b52b5e35aa9a9800c1f7dfe189437ce43253
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33342}
