When disabled, the test ResolutionAdaptsToAvailableBandwidth fails when
using frame buffer3. It is not clear if that is a problem with the test
or if that behaviour is required, and thus it is safer to have this
enabled by default and experiment with turning it off in the future.
Change-Id: I7a6ae14c37a0cdc3e203f39f6cc0c3ad87038a60
Bug: webrtc:13343
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247700
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35764}
Currently `CreateLibaomAv1Encoder` will either return an actual libaom AV1 encoder or a nullptr depening on whether the build flag `enable_libaom` was configured to true or not. This CL updates the `libaom_av1_encoder` build target to no longer depend on `enable_libaom` so that `CreateLibaomAv1Encoder` will always return an encoder instance.
Added `CreateLibaomAv1EncoderIfSupported` as a replacement to the old `CreateLibaomAv1Encoder`.
Bug: webrtc:13573
Change-Id: Ibdcd52c609acd79feefa2b86f19d1b4ca3e91d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35763}
A number of utility functions to be shared between frame buffer 2
and the new frame scheduling implementation based on frame buffer 3.
Change-Id: Icc932c6c76fddeeedc8aa64ec27c7e0c955abfd0
Bug: webrtc:13343
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241604
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35743}
This CL updates both the static GOF pattern with the correct flags for
temporal_up_switch, as well the flexible mode logic to base the flag
on dependency descriptors instead use reference buffers.
Bug: webrtc:13576
Change-Id: I578f744bec51d1f3531da5f4a89d12f05a16a6c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247187
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35741}
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}
the new spelling is more standard and more compact, in particular doesn't need extra include and thus dependency
Bug: None
Change-Id: Iaea69d2154e4d9eff2468514f5734cb3fe016ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35709}
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}
Currently some RTPVideoHeaders are not filled with width and height
information, such as AV1. If we dump the stream with command line
“--force-fieldtrials=WebRTC-DecoderDataDumpDirectory/./”, and if
width and height are 0, it will crash soon.
This CL aims to avoid crashing when the |encoded_image._encodedWidth|
and |encoded_image._encodedHeight| are 0.
Bug: webrtc:13491
Change-Id: Ie5af58c03f09a9784ed67943dc5b5959850b4368
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242500
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35576}
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}
kIvfHeaderSize is defined both inside of ivf_file_writer.cc and
ivf_file_reader.cc. This patch moves its definition into a header.
Bug: webrtc:13463
Change-Id: Ia6b2fcc3434f69a1e30a7dae7bf0c90547f11d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239722
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35540}
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}
FrameBuffer3 keep track of order, decodability and continuity of the inserted frames. Compared to FrameBuffer2 which schedule frames for decoding and is thread safe, FrameBuffer3 does not schedule decoding and is thread unsafe.
Change-Id: Ic3bd540c4f69cec26fce53a40425f3bcd9afe085
Bug: webrtc:13343
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238985
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35494}
This was a remenant leftover from a previous design, which was no longer
valid after the switch to TaskQueues. ReturnReason::kStopped was not
used at all, and so Timeout or FrameFound can be inferred from whether
the frame is null or not.
Bug: webrtc:13343, webrtc:13346
Change-Id: Ib0f847b1e1192e32ea11208e48f5a3892703521e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239651
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35490}
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}
* Clearing while waiting for a frame should return a new frame
entering the buffer.
* Stopping while waiting for a frame should cancel the wait.
Bug: webrtc:13343
Change-Id: Ife9abfa8b6ea56141c9f32ff37d3b2a2e62a44f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236849
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35314}
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}
Field trial is not used in any rollouts and should be removed.
R=mhoro@webrtc.org
Bug: webrtc:13264
Change-Id: Ib896dcdec81db7c3f4e68a8dda266d96dfdc6aed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234865
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35195}
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}
The |slice_qp_detla| reported by the hardware is not credible, which
causing the quality scaler cannot work properly,the resolution cannot
be adjusted correctly.
To fix this issue, this CL implements a bandwidth scaler which is used
for adjust resolution, this scaler will be used when QP based quality
scaler is not working due to untrusted QP reported by HW AVC encoder.
Bug: webrtc:12942
Change-Id: I2fc5f07a5400ec7e5ead2c2c502faee84d7f2a76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35120}
min_pacing:8ms, to avoid the situation where bursts of frames are sent
to the decoder at once due to network jitter. The bursts of frames
caused the queues further down in the processing to be full and
therefore drop all frames.
max_decode_queue_size:8, in the event that too many frames have piled
up, do as before and send all frames to the decoder to avoid building
up any latency.
These setting only affect the low-latency video pipeline that is enabled
by setting the playout RTP header extension to min=0ms, max>0ms.
Bug: chromium:1138888
Change-Id: I8154bf3efe7450b770da8387f8fb6b23f6be26bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233220
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35119}
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}
The VP9 encoder may drop a frame internally which will not advance the
frame pattern. Consider the following scenario where only spatial layer
0 and temporal layer 0 is active:
1. Key frame encoded
2. Spatial layer 1 is activated
3. Delta T0 dropped
4. Delta T0 encoded
No S1T0 frame is encoded in (1) since it's not active. When
NextFrameConfig is called in (3) it will say that future frames may
reference T0 on both S0 and S1, but it's then dropped.
On step (4), the SVC controller essentially thinks it's encoding a new
picture and will happily reference the T0 on what it thinks is the first
delta frame. However, this is actually still the key frame and since
there was no S1T0 frame produced it will reference an invalid buffer.
To fix this, only say it's possible to reference a T0 frame after it has
been successfully encoded.
Bug: webrtc:11999, webrtc:13142, chromium:1178444
Change-Id: Iab3d2042ce0b3fa7d952b2831d1a36b1a6613a86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231695
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34982}
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}
delta_q is encoded as signed integer (s(4)) that uses extra bit for the
sign. See VP9 Bitstream Specification section 6.2.10 Delta quantizer syntax
Bug: None
Change-Id: Ib458c2a2ded3c4d6c153b6bedd29c48ef12cc538
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231125
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34908}
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}
from runtime check in proxy classes that picks decoder (VCMDecoderDataBase)
to a DCHECK in the VideoDecoder::Settings
Bug: None
Change-Id: Ic8c2e971486a3a7eb247f9d03815aba5ca5a7bad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228644
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34761}
As a first step we only want to enable frame pacing for the case
where min playout delay == 0 and max playout delay > 0.
Bug: chromium:1237402, chromium:1239469
Change-Id: Icf9641db7566083d0279135efa8618e435d881eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228640
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34752}
When pacing is enabled for the low latency rendering path,
frames are sent to the decoder in regular intervals. In case of a
jitter, these frames intervals could add up to create a large latency.
Hence, disable frame pacing if the pre-decode queue grows beyond the
threshold. The threshold for when to disable frame pacing is set
through a field trial. The default value is high enough so that
the behavior is not changed unless the field trial is specified.
Bug: chromium:1237402
Change-Id: I901fd579f68da286eca3d654118f60d3c55e21ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228241
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34705}
Remove private members that are no longer used or always have same value
Use less allocations
Bug: None
Change-Id: I5430c2356f0039212baf8b248b92775e8852ce1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227765
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34665}
Schedule the frames to be decoded based on the pacing delay from the
last decode scheduled time. In the current implementation, multiple
threads and different functions in same thread can call
MaxWaitingTime(), thereby increasing the wait time each time the
function is called. Instead of returning the wait time for a future
frame based on the number of times the function is called, return the
wait time only for the next frame to be decoded. Threads can call the
function repeatedly to check the waiting time for next frame and wake
up and then go back to waiting if an encoded frame is not available.
Change-Id: I00886c1619599f94bde5d5eb87405572e435bd73
Bug: chromium:1237402
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226502
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34660}
The alternative new name proposed, NackTracker, is already in
use in audio_coding.
Fixed: webrtc:11594
Change-Id: I6a05fafc05fa7ddb18ea4f64886a135e5ef59f7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226744
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34539}
As part of go/coil update code search links to not point to the
"master" branch.
Bug: chromium:1226942
Change-Id: I0ae9e84ecc660f789a69fe0b226f93bbc39a8a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226081
Commit-Queue: Tony Herre <toprice@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34531}
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}
NackModule2 creates repeating tasks, but as there are
many modules (one per receiver) these tasks execute out
of phase with each other, multipliying the amount of wakeups
caused.
Fix this by creating a single wakeup source that serves all
NackModule2 instances in a call.
Bug: webrtc:12989
Change-Id: Ia9c84307eb57349679e42b673474feb2cb43f08e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226464
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34527}
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}
