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}
Addresses case where 540*960 would not get a 135*240 layer.
Bug: webrtc:13469
Change-Id: Icc291c65114fb400cc71659d76a786e359e5996c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239820
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Konrad Hofbauer <hofbauer@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35507}
This allows to differentiate and test codecs of the same type but
different implementations/settings.
Bug: none
Change-Id: I74f799b36411e63387513133ffc19a7f0c45d550
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238165
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35396}
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 CL set the spatial id in LibaomAv1Encoder and set correct number
of spatial layers for AV1 in FrameEncodeMetadataWriter.
Bug: None
Change-Id: I40092e45be88ec9ab75f228d9ca84c44e3cad326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237662
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Cr-Commit-Position: refs/heads/main@{#35339}
This adds the Main 3.1 profile to the list of supported H264 codecs. This unifies the output of WebRTC codecs among macOS/Windows (which both have Main 3.1 codecs) and headless Linux browsers.
Bug: None
Change-Id: Ife2fe8c1827be9368fabccc5f24ba316671b1b8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236600
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35328}
This ensures that the payload descriptor and potential generic
descriptors uses the same temporal layer.
Bug: b/200518293
Change-Id: I17e980b47fe6c814cb393fc459064576447aa27a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236520
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35275}
It seems the Android CTS tests only verify that 16x16 aligned resolutions
are supported.
This change checks the validity of input frame's size when initialing
or encoding processes are about to start using H/W MediaCodec.
This change has additional APIs to retrieve
|requested_resolution_alignment| and |apply_alignment_to_all_simulcast_layers|
from JAVA VideoEncoder class and its inherited classes. HardwareVideoEncoder
using MediaCodec has values of 16 and true for above variables.
Bug: webrtc:13089
Change-Id: I0c4ebf94eb36da29c2e384a3edf85b82e779b7f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229460
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35169}
All valid scalability modes should be supported by the builtin
software decoder/encoder.
Bug: chromium:1187565
Change-Id: If66105d210d5055019f35dae2f80a18ad4a70cdd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222642
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34998}
Unlike libvpx, the VideoBitrateAllocation expects that the bitrate
allocation is separate for each temporal layer. In this instance, if the
bitrates are not separated it will fool the SVC controller into thinking
that all temporal layers are always active.
Bug: webrtc:11999
Change-Id: Ibc33ac00b8b7716c011b94e1ec9c640cedb5274e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231693
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34980}
This cover scenario where target bitrate is changed in a middle of
of group of frame after spatial upswitch.
This change should avoid wasting encoder resources to produce those
frames, reduce number of errors
"Encoder produced a frame for layer that wasn't requested"
Bug: webrtc:11999
Change-Id: I06045259b1cee2c21bfdabbafff3892b57c82a84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230543
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34969}
This is a reland of 3097008de0
Patchset 1 is a pure reland. Patchset 2 contains a bugfix plus a test
covering that case.
Bug: webrtc:12354, chromium:1230448
Original change's description:
> Rename vp9::FrameInfo to vp9::UncompressedHeader and add more fields.
>
> These fields will be used for bitstream validation in upcoming CLs.
> A new vp9_constants.h file is also added, containing common constants
> defined by the bitstream spec.
>
> Bug: webrtc:12354
> Change-Id: If04256d83409069c8bee43ad41aed41c3707dfd3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226060
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34476}
Bug: webrtc:12354
Change-Id: Ibd301eb458a6104b562cefbc0e616c39b54fb38b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229060
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34789}
This CL completes the removal of assert() and relative headers from
the codebase (excluded
//examples/objc/AppRTCMobile/third_party/SocketRocket which is in a
third_party sub-directory).
Bug: webrtc:6779
Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34528}
This reverts commit 3097008de0.
Reason for revert: suspected crash
Bug: chromium:1230239
TBR=philipel@webrtc.org
Original change's description:
> Rename vp9::FrameInfo to vp9::UncompressedHeader and add more fields.
>
> These fields will be used for bitstream validation in upcoming CLs.
> A new vp9_constants.h file is also added, containing common constants
> defined by the bitstream spec.
>
> Bug: webrtc:12354
> Change-Id: If04256d83409069c8bee43ad41aed41c3707dfd3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226060
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34476}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12354
Change-Id: Ia4d5180d593c66a053d5747e714a579c62ea2a37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226327
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34496}
These fields will be used for bitstream validation in upcoming CLs.
A new vp9_constants.h file is also added, containing common constants
defined by the bitstream spec.
Bug: webrtc:12354
Change-Id: If04256d83409069c8bee43ad41aed41c3707dfd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226060
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34476}
iNumRefFrame specifies total number of reference buffers to allocate.
For N temporal layers we need at least (N - 1) buffers to store last
encoded frames of all reference temporal layers.
There is no API in OpenH254 encoder to specify exact set of references
to be used to prediction of a given frame. Encoder can theoretically
use all available references.
Note that there is logic in OpenH264 which overrides iNumRefFrame to
max(iNumRefFrame, N - 1): https://source.chromium.org/chromium/chromium/src/+/main:third_party/openh264/src/codec/encoder/core/src/au_set.cpp;drc=8e90a2775c5b9448324fe8fef11d177cb65f36cc;l=122.
I.e., this change has no real effect. It only makes setup more clear.
Bug: none
Change-Id: If4b4970007e1cc55d8f052ea05213ab2e89a878f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225480
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34445}
CL partially auto-generated with:
git grep -l "\bassert(" | grep "\.[c|h]" | \
xargs sed -i 's/\bassert(/RTC_DCHECK(/g'
And with:
git grep -l "RTC_DCHECK(false)" | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'
With some manual changes to include "rtc_base/checks.h" where
needed.
A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.
The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.
This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).
Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
In cases where ToI420 fails it should be able to return null.
Bug: webrtc:12877
Change-Id: Ia13859c104d978a29712ae10f8e15acada8406ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222613
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34342}
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}
If at creation of a VP8 encoder there is not enough bitrate to enable a
given spatial layer - the configuration won't be updated to indicate
the correct temporal layer count. This means GetEncoderInfo() will
indicate lack of temporal layer support, which triggers issues with
rate allocation.
This CL fixes that by always setting an initial bitrate of 0bps.
Bug: webrtc:12788
Change-Id: I10974e85446b58e597d2ca415eaf2550306ce986
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220929
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34198}
ffmpeg is going to be hiding the implementation of AVPacket, so we can't
allocate them on the stack anymore. av_init_packet is marked deprecated
on TOT ffmpeg, so remove its use everywhere in favor of av_packet_alloc
and av_packet_free.
Bug: chromium:1211508
Change-Id: I154311071123110dd749c71dec1ec2a0452b3908
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217780
Commit-Queue: Ted Meyer <tmathmeyer@google.com>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34106}
In CL https://webrtc-review.googlesource.com/c/src/+/216323 we fixed
the issue where I420 and I420A not being equal would result in dropping
frames in release builds.
But we forgot to update the corresponding DCHECK, meaning the I420 not
being the same as I420A issue still causes crashes on debug builds.
(I must have been running a release build not to catch this before?)
This CL replaces the DCHECK_EQ with an RTC_NOTREACHED inside the
IsCompatibleVideoFrameBufferType check.
Because this only affects debug builds, this CL does not need to be
backmerged anywhere.
Bug: chromium:1203206
Change-Id: I101823e8bca293e94d0f7ce507fe78cedff3ea1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219281
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34048}
When VideoFrameType for svc upper layer is kVideoFrameDelta for key pic,
the svc unittest will fail due to the wrong frame type for the super
frame of first key picture.
Bug: None
Change-Id: Iff026aaecb73890d3c45d2c88c9654a12d6fe3bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216461
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Cr-Commit-Position: refs/heads/master@{#33986}
VideoCodecInitializer::SetupCodec never sets startBitrate,
so SetAv1SvcConfig shouldn't use it.
Bug: webrtc:12720
Change-Id: I04835dc27368f32c19132d93c72364173d7050fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217382
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33915}
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.
Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
In Chromium, the I420ABufferInterface implementation uses the default
CropAndScale() implementation which converts to I420 in the process.
This should be OK, because we do not encode the alpha channel anyway,
so having WebRTC scaling ignore the alpha channel might even be a good
thing. Unfortunatety, an if statement in the LibvpxVp8Encoder did not
consider I420A and I420 to be the same, resulting in dropping perfectly
valid frames.
This CL fixes that by considering I420A and I420 "compatible" in a
comparison helper function. The problem only happens in this encoder,
so only this encoder needs to be fixed.
Bug: chromium:1203206
Change-Id: Iec434d4ada897c79e09914cac823148fd5b05e57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216323
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33845}
This is a refactor to simplify a follow-up CL of adding
SdpVideoFormat::IsSameCodec.
The original files media/base/h264_profile_level_id.* and
media/base/vp9_profile.h must be kept until downstream projects
stop using them.
Bug: chroimium:1187565
Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33782}
...though the big issue was probably that pending frames weren't being
culled properly in the case of frame dropping.
Bug: webrtc:12596
Change-Id: I9a03282b2a99087aa7c5650e57ce30fe0f0d3036
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214127
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33638}
While debugging https://crbug.com/1195144 I found it useful to clarify
this log statement.
The log would say "When scaling [kNative], the image was unexpectedly
converted to [kI420]..." but not saying what it was trying to convert
it to. This CL adds: "... instead of [kNV12]."
Bug: chromium:1195144
Change-Id: I13e0040edf5d7d98d80ce674812f67dfb73be36e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214040
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33634}
libaom is compiled with REALTIME_ONLY option. Soon it will be impossible
to create encoder or request default config with usage other than
AOM_USAGE_REALTIME. Fixing the wrapper to use proper usage parameter
Bug: None
Change-Id: I862741a724e4a8524f22ae79700b3da6517dbfb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214100
Commit-Queue: Fyodor Kyslov <kyslov@google.com>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33624}
Follow-up CL to VP8 and VP9 encoders taking care of mapping.
Context again:
This CL is part of Optimized Scaling efforts. In Chromium, the native
frame buffer is getting an optimized CropAndScale() implementation. To
support HW accelerated scaling, returning pre-scaled images and skipping
unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale
instead of libyuv::XXXXScale and 2) only map buffers it actually intends
to encode.
In this CL, VideoStreamEncoder no longer calls GetMappedFrameBuffer() on
behalf of the encoders, since the encoders are now able to either do the
mapping or performs ToI420() anyway.
- Tests for old VSE behaviors are updated to test the new behavior (i.e.
that native frames are pretty much always forwarded).
- The "having to call ToI420() twice" workaround to Android bug
https://crbug.com/webrtc/12602 is added to H264 and AV1 encoders.
Bug: webrtc:12469
Change-Id: Ibdc2e138d4782a140f433c8330950e61b9829f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211940
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#33548}
This will further speed up intra frame encoding
Bug: None
Change-Id: I3c836502cdcb1037e3128850a085b92acd8fc7ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212821
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Fyodor Kyslov <kyslov@google.com>
Cr-Commit-Position: refs/heads/master@{#33544}
This is a follow-up to the VP9, fixing VP8 this time. Context again:
This CL is part of Optimized Scaling efforts. In Chromium, the native
frame buffer is getting an optimized CropAndScale() implementation. To
support HW accelerated scaling, returning pre-scaled images and skipping
unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale
instead of libyuv::XXXXScale and 2) only map buffers it actually intends
to encode.
- To achieve this, WebRTC encoders are updated to map kNative video
buffers so that in a follow-up CL VideoStreamEncoder can stop mapping
intermediate buffer sizes.
Bug: webrtc:12469, chromium:1157072
Change-Id: I026527ae77e36f66d02e149ad6fe304f6a8ccb05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212600
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#33537}
This CL is part of Optimized Scaling efforts. In Chromium, the native
frame buffer is getting an optimized CropAndScale() implementation. To
support HW accelerated scaling, returning pre-scaled images and skipping
unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale
instead of libyuv::XXXXScale and 2) only map buffers it actually intends
to encode.
- To achieve this, WebRTC encoders are updated to map kNative video
buffers so that in a follow-up CL VideoStreamEncoder can stop mapping
intermediate buffer sizes.
In this CL LibvpxVp9Encoder is updated to map kNative buffers of pixel
formats it supports and convert ToI420() if the kNative buffer is
something else. A fake native buffer that keeps track of which
resolutions were mapped, MappableNativeBuffer, is added.
Because VP9 is currently an SVC encoder and not a simulcast encoder, it
does not need to invoke CropAndScale.
This CL also fixes MultiplexEncoderAdapter, but because it simply
forwards frames it only cares about the pixel format when
|supports_augmented_data_| is true so this is the only time we map it.
Because this encoder is not used with kNative in practise, we don't care
to make this path optimal.
Bug: webrtc:12469, chromium:1157072
Change-Id: I74edf85b18eccd0d250776bbade7a6444478efce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212580
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#33526}
This will further speed up intra frame encoding
Bug: None
Change-Id: I1a105c6d2cdd9dc82f84d0039dbea3f0d090ab93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212320
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Fyodor Kyslov <kyslov@google.com>
Cr-Commit-Position: refs/heads/master@{#33492}
This will speed up key frame encoding (together with libaom changes)
3x-4x times with ~13% BDRate loss on key frames only
Bug: None
Change-Id: I24332f4f7285811cdc6619ba29844fe564cae95e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212040
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Fyodor Kyslov <kyslov@google.com>
Cr-Commit-Position: refs/heads/master@{#33468}
Check if codec was successfully created and exit from RunTest if not
before creating VideoProcessor.
Bug: none
Change-Id: Ia6d7171650dbc9824fb78f4a8e2851f755cfd63b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209362
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33372}
Use 4 threads for 360p and above.
Use tile rows for VGA and 4 threads.
Use speed 8 for 360p.
Change min max qp scaling threshold.
Bug: None
Change-Id: Ib7a5b7e539d26d9fa60aa2c4a75eb6f4b19f7dea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208340
Commit-Queue: Jerome Jiang <jianj@google.com>
Commit-Queue: Fyodor Kyslov <kyslov@google.com>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33320}
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}
Keeping structures in the same file makes it clearer which are missing
and makes it easier to see if structures are consistent with one another.
No-Try: True
Bug: None
Change-Id: I4e5e6971054dd28dd326c68369ee57b6df62725e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206987
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33256}
Since for such frame SvcController haven't setup how buffer should be
referenced and updated, the frame would likely have unexpected configuration.
Log an error to note resource have been wasted produce it and drop such frame.
Bug: webrtc:11999
Change-Id: I1784403e67b7207092d46016510460738994404e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205140
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33148}
To be able to build these targets in chromium we need to replace all abseil dependencies with "//third_party/abseil-cpp:absl".
Bug: webrtc:12404
Change-Id: Ie0f6af73f2abc73e5744520cfd9a6414e2f948e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202762
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33108}
To make VideoCodec::scalability_mode the only option to set and
change the scalability structure, for easier maintainability.
Bug: webrtc:11404
Change-Id: I6570e9a93ddf2897ff7584c5d20a246346e853e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192361
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33056}
Using WebRTC-VP9-PerformanceFlags and settings a multi-layer config,
and then configuring the codec in non-svc mode would cause us to not
set the cpu speed in libvpx. For some reason, that could trigger a
crash in the encoder.
This CL fixes that, and adds new test coverage for the code affected
byt the trial.
Bug: chromium:1167353, webrtc:11551
Change-Id: Iddb92fe03fc12bac37717908a8b5df4f3d411bf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202761
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33051}
This adds a new way to poll decoder metadata.
A default implementation still delegates to the old methods.
Root call site is updates to not use the olds methods.
Follow-ups will dismantle usage of the olds methods in wrappers.
Bug: webrtc:12271
Change-Id: Id0fa6863c96ff9e3b849da452d6540e7c5da4512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196520
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32976}
Absent fps allocation imply single layer stream which confuses bitrate adjuster.
As a result bitrate adjuster turned off S0T1 and S0T2 layers for the L3T3 structure.
Bug: webrtc:12148
Change-Id: I5b3a7b44322f347f41dd8858b3d703827e69dd72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201384
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32952}
The rtc::Bind usages are replaced with lambdas with copy-capture
of the ref pointers.
Bug: webrtc:11339
Change-Id: I2fb544fcd2780feac3d725993c360df91899b532
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32946}
Lowering a bit since it is currently failing after one of CLs from https://aomedia.googlesource.com/aom.git/+log/87c414ed32..43927e4611
The error is "error: Expected: (video_stat.min_ssim) > (quality_thresholds->min_min_ssim), actual: 0.919629 vs 0.92"
Bug: None
Change-Id: I35e1e989961c6794a7f5f2015f5a8a786f1e25f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197808
Reviewed-by: Jerome Jiang <jianj@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32844}
Adds ability to specify desired frame size separate from actual clip
resolution, as well as clip and desired fps.
This allows e.g. reading an HD clip but running benchmarks in VGA, and
to specify e.g. 60fps for the clip but 30for encoding where frame
dropping kicks in so that motion is actually correct rather than just
plaing the clip slowly.
Bug: webrtc:12229
Change-Id: I4ad4fcc335611a449dc2723ffafbec6731e89f55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195324
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32839}
Also moves the LibvpxVp8Interface from codec/vp8 to codec/interface and
drops vp8 from the name.
Follow-up CLs will wire up actual usage in the new classes through the
interface so that we can unit test them more easily.
Bug: webrtc:12274
Change-Id: I95f66e90245d9320e5fc23cdc845fbeb2648b38b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196522
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32816}
This is a reland of 6e7167456b
Patch set 1 is the original.
Later patch sets fix a parsing bug, and adds a new flag which enables
or disabled the ability to set separate per spatial layer speed
(use_per_layer_speed).
Original change's description:
> Adds experimental libvpx VP9 speed settings.
>
> Using the field trial WebRTC-VP9-PerformanceFlags, this CL allows you to
> configure the libvpx VP9 encoder with a list of flags to affect the
> quality vs speed tradeoff. This CL adds support for:
>
> * Speed (effort), for the temporal base layer frames
> * Speed for higher (non-base) layer frames
> * De-blocking (as part of the loopfilter) enabled for:
> 0 = all frames
> 1 = all but frames from the highest temporal layer
> 2 = no frames
>
> Each entry in the list has a threshold in min number of pixels needed
> for settings in the entry to apply.
>
> Example: Two spatial layers (180p, 360p) with three temporal
> layers are configured. Field trial "WebRTC-VP9-PerformanceFlags" set to:
> "use_per_layer_speed,min_pixel_count:0|129600,base_layer_speed:5|7,high_layer_speed:8|8,deblock_mode:1|2"
> This translates to:
> S0:
> - TL0: Speed 5, deblocked
> - TL1: Speed 8, deblocked
> - TL2: Speed 8, not deblocked
> S1:
> - TL0: Speed 7, not deblocked
> - TL1: Speed 8, not deblocked
> - TL2: Speed 8, not deblocked
>
> Bug: webrtc:11551
> Change-Id: Ieef6816d3e0831ff53348ecc4a90260e2ef10422
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188461
> Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32749}
Bug: webrtc:11551
Change-Id: Ie7c703eb122197235d8ce77cb72db7a347382468
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196345
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32780}
This reverts commit 6e7167456b.
Reason for revert: Unexpected perf change
Original change's description:
> Adds experimental libvpx VP9 speed settings.
>
> Using the field trial WebRTC-VP9-PerformanceFlags, this CL allows you to
> configure the libvpx VP9 encoder with a list of flags to affect the
> quality vs speed tradeoff. This CL adds support for:
>
> * Speed (effort), for the temporal base layer frames
> * Speed for higher (non-base) layer frames
> * De-blocking (as part of the loopfilter) enabled for:
> 0 = all frames
> 1 = all but frames from the highest temporal layer
> 2 = no frames
>
> Each entry in the list has a threshold in min number of pixels needed
> for settings in the entry to apply.
>
> Example: Two spatial layers (180p, 360p) with three temporal
> layers are configured. Field trial "WebRTC-VP9-PerformanceFlags" set to:
> "min_pixel_count:0|129600,base_layer_speed:5|8,high_layer_speed:7|8,deblock_mode:1|2"
> This translates to:
> S0:
> - TL0: Speed 5, deblocked
> - TL1: Speed 8, deblocked
> - TL2: Speed 8, not deblocked
> S1:
> - TL0: Speed 7, not deblocked
> - TL1: Speed 8, not deblocked
> - TL2: Speed 8, not deblocked
>
> Bug: webrtc:11551
> Change-Id: Ieef6816d3e0831ff53348ecc4a90260e2ef10422
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188461
> Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32749}
TBR=sprang@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org
Change-Id: If910963441ac1a0e002aac7066791c7cc7764a1a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11551
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196344
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32762}
Using the field trial WebRTC-VP9-PerformanceFlags, this CL allows you to
configure the libvpx VP9 encoder with a list of flags to affect the
quality vs speed tradeoff. This CL adds support for:
* Speed (effort), for the temporal base layer frames
* Speed for higher (non-base) layer frames
* De-blocking (as part of the loopfilter) enabled for:
0 = all frames
1 = all but frames from the highest temporal layer
2 = no frames
Each entry in the list has a threshold in min number of pixels needed
for settings in the entry to apply.
Example: Two spatial layers (180p, 360p) with three temporal
layers are configured. Field trial "WebRTC-VP9-PerformanceFlags" set to:
"min_pixel_count:0|129600,base_layer_speed:5|8,high_layer_speed:7|8,deblock_mode:1|2"
This translates to:
S0:
- TL0: Speed 5, deblocked
- TL1: Speed 8, deblocked
- TL2: Speed 8, not deblocked
S1:
- TL0: Speed 7, not deblocked
- TL1: Speed 8, not deblocked
- TL2: Speed 8, not deblocked
Bug: webrtc:11551
Change-Id: Ieef6816d3e0831ff53348ecc4a90260e2ef10422
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188461
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32749}
so that it will be filled in the dependency descriptor rtp header extension
Bug: webrtc:10342
Change-Id: Ifaf4963ca84f6d495287959746686ae3dcd176d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168767
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32692}
This is a reland of f5e261aaf6
This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since
there have been some issues.
Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}
Bug: webrtc:11495, webrtc:12218
Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32668}
In particular move end_of_picture flag out of vp9 specific information
since VP9 is not the only codec that can use spatial scalability and
thus need to distinguish layer frame and picture (aka temporal unit).
Bug: webrtc:12167
Change-Id: I0d046d8785fbea55281209ad099738c03ea7db96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192542
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32588}
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.
Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
After recently changing .pylintrc (see [1]) we discovered that
the presubmit check always checks all the python files when just
one python file gets updated.
This CL moves all these files one step closer to what the linter
wants.
Autogenerated with:
# Added all the files under pylint control to ~/Desktop/to-reformat
cat ~/Desktop/to-reformat | xargs sed -i '1i\\'
git cl format --python --full
This is part 1 out of 2. The second part will fix function names and
will not be automated.
[1] - https://webrtc-review.googlesource.com/c/src/+/186664
No-Presubmit: True
Bug: webrtc:12114
Change-Id: Idfec4d759f209a2090440d0af2413a1ddc01b841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32530}
Calculate quality metrics for dropped frames by comparing original
frame against last decoded one.
This feature makes comparison of encoders which do/don't drop frames
more fair.
The feature is controlled by analyze_quality_of_dropped_frames flag
and is disabled by default.
Bug: none
Change-Id: Ifab8df92d0b76e743ff3193c05d7c8dbd14921c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190660
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32518}
In the old jitter buffer the two VCMVideoProtection modes |kProtectionNone| and |kProtectionFEC| could be set on the jitter buffer for it to not wait for NACK and instead generate incomplete frames. This has not been possible for a long time.
Bug: webrtc:9378, webrtc:7408
Change-Id: I0a2d3ec34d721126c1128306d5fad88314f8d59f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32513}
INCREASING_ID, which is the default mode, triggers HW reset in chromium
decoder wrapper. Set eSpsPpsIdStrategy to SPS_LISTING to prevent that.
Note that WebRTC always resets the encoder on resolution change. This
makes all strategies except INCREASING_ID essentially equivalent to
CONSTANT_ID.
Bug: chromium:1111273
Change-Id: I37405c97b3390f812d1dcaa111694b3b1d638035
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190440
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32505}
"warning: control reaches end of non-void function [-Wreturn-type]"
Reported by gcc (8.3)
In all the reported cases, the end of function is never actually
reached. Add RTC_CHECK(false) to ensure the compiler is aware that
this path is a dead-end.
Bug: webrtc:12008
Change-Id: I7f816fde3d1897ed2774057c7e05da66e1895e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fabien VALLÉE <fabien.vallee@netgem.com>
Cr-Commit-Position: refs/heads/master@{#32503}
This changes the default behavior to use pacing factor of 1.1x instead
of 2.5x, it also sets libvpx rate controler as trusted, turns on the
encoder pushback mechanism and sets spatial hysteresis to 1.2.
The unused "dynamic rate" settings in libvpx is removed.
The new settings matches what has been used in chromium since 2019.
If needed, the legacy behavior can be enabled using the field trial
WebRTC-VideoRateControl.
Bug: webrtc:10155
Change-Id: I8186b491aa5bef61e8f568e96c980ca68f0c208f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186661
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32477}
To make it natural to reuse them for vp9
Bug: webrtc:11999
Change-Id: If2ef7ca16b8be96e0e03bb19211d9f5eb74b2d3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188620
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32414}
to make it a bit simpler
Bug: None
Change-Id: Ie6288594c5a1b8535007623032b422eefc716ca6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188460
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32405}
This is a reland of 76d3e7a8d1
I have run the WPT tests and ensured they are now passing with this
change. I have changed the following,
- The old CL was assuming that ToI420 frames had type I420, but they
could be I420A which was causing a crash.
- I fixed a copy-paste error in the offset of the V stride.
Original change's description:
> NV12 support for VP8 simulcast
>
> Tested using video_loopback with generated NV12 frames.
>
> Bug: webrtc:11635, webrtc:11975
> Change-Id: I14b2d663c55a83d80e48e226fcf706cb18903193
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186722
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32325}
Bug: webrtc:11635
Bug: webrtc:11975
Change-Id: Ifa790af97cd7ab30c6cb4648ebd140abc1593b0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187490
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32381}
(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}
Adds a field to EncoderInfo called preferred_pixel_formats which a
software encoder populates with the pixel formats it supports. When a
kNative frame is received for encoding, the VideoStreamEncoder will
first try to get a frame that is accessible by the software encoder in
that pixel format from the kNative frame. If this fails it will fallback
to converting the frame using ToI420.
This minimizes the number of conversions made in the case that the
encoder supports the pixel format of the native buffer or where
conversion can be accelerated. For example, in Chromium, the capturer can
emit an NV12 frame, which can be consumed by libvpx which supports NV12.
Testing: Tested in Chrome with media::VideoFrame adapters.
Bug: webrtc:11977
Change-Id: I9becc4100136b0c0128f4fa06dedf9ee4dc62f37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187121
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32353}
