This emulates behaviour from frame buffer 2, but does not handle stats.
In contrast to frame buffer 2, all work happens on the same task queue.
FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind
a field trial WebRTC-FrameBuffer3.
This separates frame scheduling behaviour into a few components,
VideoReceiveStreamTimeoutTracker
* Handles the stream timeouts.
FrameDecodeScheduler
* Manages the scheduling and cancelling of frames being sent to the
decoder.
FrameDecodeTiming
* Handles the timing and ordering of frames to be decoded.
Other changes
* Adds CurrentSize() method to FrameBuffer3
* Move timing to a separate library
* Does a thread check for Receive statistics as this is now
on the worker thread.
* Adds `FlushImmediate` method to RunLoop so that
video_receive_stream2_unittest can pass when scheduling is happening
on the worker thread.
Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721
Bug: webrtc:13343
Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35847}
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}
This is a delegate that is used by video_receive_stream2 to handle frame
buffer tasks like threading, and stats. This will be used in a follow up
to use FrameBuffer3 as a strategy selected by field trial.
Unit-tests will be used in follow-up CLs containing Frame Buffer 3, and
are expected to work with both Frame buffer proxy versions.
Change-Id: I524279343d60a348d044d9085d618f12d7bf3a23
Bug: webrtc:13343
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241605
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@{#35803}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
This unblocks lowering the precision of low precision tasks which are
the default.
Bug: webrtc:13604
Change-Id: Icd663cbbf5b0bf87ac83a4a0abd58699e6e27e8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248862
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35782}
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}
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}
If the RTP header extension playout-delay is used and set
to min=0, max>=0, frames are scheduled to be decoded as
soon as possible. There's a risk that too many frames are
sent to the decoder at once, which may cause problems
further down in the video pipeline.
This CL adds the fieldtrial WebRTC-ZeroPlayoutDelay with
the parameter min_pacing that determines the minimum
pacing interval between two frames scheduled for
decoding.
Bug: None
Change-Id: I471f7718761cfce9789b3aa8adea3e8a16ecb2fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223742
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34387}
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}
Specifically, use reference instead of pointer for out parameter
and place the out parameter last, for the following methods
ReadUInt8
ReadUInt16
ReadUInt32
ReadBits
PeekBits
ReadNonSymmetric
ReadSignedExponentialGolomb
ReadExponentialGolomb
Bug: webrtc:11933
Change-Id: I3f1efe3e29155985277b0cd18700ddea25fe7914
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218504
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34037}
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}
Historically the PacketBuffer used a callback for assembled frames, and because of that RtpPacketInfos were piped through it even though they didn't have anything to do with the PacketBuffer.
With this CL RtpPacketInfos are stored in the RtpVideoStreamReceiver(2) instead.
Bug: webrtc:12579
Change-Id: Ia6285b59e135910eee7234b89b23425bb0fc0d2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215320
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33980}
Before this CL, timestamps of received packets were rounded
to the nearest millisecond and stored as int64_t. Due to the
rounding it sometimes happened that timestamps later in the
pipeline that are not rounded seem to occur even before the
video frame was received.
Change-Id: I92d8f3540b23baae2d4a1dc6a7cb3f58bcdaad18
Bug: webrtc:12722
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216398
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33916}
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}
Also fix similar annotation on NackModule to have effect
(when defining an alias with C++ using, ABSL_DEPRECATED should appear
on the left hand side).
Bug: webrtc:12339
Change-Id: Id80a20bf2c56a826777b8a40e06ac5c65e4f8db7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217242
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33905}
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}
Fix simulcast svc controller to reuse dropped frame configuration,
same as full svc and k-svc controllers do.
This fuzzer reminded the issue was still there.
This is a reland of https://webrtc-review.googlesource.com/c/src/+/212281
Bug: webrtc:11999
Change-Id: Id3b2cd6c7e0923adfffb4e04c35ed2d6faca6704
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33802}
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}
Biggest change is a new helper class used to read data from the
bitstream and then pass the result to a function if reading was
successful. There's also helper to do if/else flow based on the read
values. This avoids a bunch of temporaries and in my view makes the
code esaier to read.
For example, this block:
uint32_t bit;
RETURN_FALSE_IF_ERROR(br->ReadBits(&bit, 1));
if (bit) {
RETURN_FALSE_IF_ERROR(br->ConsumeBits(7));
}
...is now written as:
RETURN_IF_FALSE(
br->IfNextBoolean([br] { return br->ConsumeBits(7); }));
In addition, we parse and put a few extra things in FrameInfo:
show_existing_frame, is_keyframe, and base_qp.
Bug: webrtc:12354
Change-Id: Ia0b707b223a1afe0a4521ce2b995437d41243c06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215239
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33776}
Like aom and openh264, VP9 can be disabled with the gn argument.
Bug: None
Change-Id: I7d67e3946afae0bb4cac8a7e591445604dda9ce1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215260
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33737}
In RtpVideoStreamReceiver2 it can be protected by the `worker_task_checker_` instead.
Bug: webrtc:12579
Change-Id: I4f7d64f16172139eddc7a3e07d1dbbf338beaf2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215224
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33734}
VideoStreamEncoderTest: Remove unneeded set_timestamp_rtp in CreateFrame methods (the timestamp is set based on ntp_time_ms in VideoStreamEncoder::OnFrame).
Bug: none
Change-Id: I6b5531a9ac21cde5dac54df6de9b9d43261e90c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214488
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33683}
There's been reports of dropped frames that are not counted and
correctly reported by getStats().
If a HW decoder is used and the system is provoked by stressing
the system, I've been able to reproduce this problem. It turns out
that we've missed frames that are dropped because there is no
callback to the Decoded() function.
This CL restructures the code so that dropped frames are counted
even in cases where there's no corresponding callback for some frames.
Bug: webrtc:11229
Change-Id: I0216edba3733399c188649908d459ee86a9093d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214783
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33671}
...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}
This change adds basic support for setting codecType kVideoCodecAV1 in
VCMEncodedFrames.
Bug: chromium:1191972
Change-Id: I258b39ff89c8b92ebbb288ef32c88b900a35d10e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213182
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33594}
This reverts commit c184047fef.
Reason for revert: Breaks the WebRTC->Chromium roll:
ERROR Unresolved dependencies.
//third_party/webrtc/test/fuzzers:vp9_encoder_references_fuzzer(//build/toolchain/win:win_clang_x64)
needs //third_party/webrtc/modules/video_coding:mock_libvpx_interface(//build/toolchain/win:win_clang_x64)
We need to add tryjob to catch these. The fix is to make
//third_party/webrtc/modules/video_coding:mock_libvpx_interface
visible in built_with_chromium builds by moving the target
out of this "if" https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/video_coding/BUILD.gn;l=615;drc=3889de1c4c7ae56ec742fb9ee0ad89657f638169.
Original change's description:
> Add fuzzer to validate libvpx vp9 encoder wrapper
>
> Fix simulcast svc controller to reuse dropped frame configuration,
> same as full svc and k-svc controllers do.
> This fuzzer reminded the issue was still there.
>
> Bug: webrtc:11999
> Change-Id: I74156bd743124723562e99deb48de5b5018a81d0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212281
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33568}
TBR=danilchap@webrtc.org,sprang@webrtc.org
Change-Id: I1676986308c6d37ff168467ff2099155e8895452
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212973
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33573}
Fix simulcast svc controller to reuse dropped frame configuration,
same as full svc and k-svc controllers do.
This fuzzer reminded the issue was still there.
Bug: webrtc:11999
Change-Id: I74156bd743124723562e99deb48de5b5018a81d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33568}
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}
Namespace used because of copy-pasting an old pattern, should never have been used in the first place. Removing it now to make followup refactoring prettier.
Bug: webrtc:12579
Change-Id: I00a80958401cfa368769dc0a1d8bbdd76aaa4ef5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212603
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33536}
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}
A refactoring (https://webrtc-review.googlesource.com/c/src/+/196520)
of decoder metadata handling introduced a bug which causes us to log an
info-level entry for every frame decoded if the implementation changes
during runtime (e.g. due to software fallback).
This CL fixes that to avoid spamming the logs.
Bug: webrtc:12271
Change-Id: I89016351b8752b259299c4cf56c6feddcca43460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211664
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33451}
The access to |_timestampMap| was guarded by a lock but
not the access to the data pointer stored in |_timestampMap|.
There was a potential race condition if new data was added
in VCMGenericDecoder::Decode() while the data pointer
retrieved from _timestampMap.Pop() was being used in
VCMDecodedFrameCallback::Decoded().
This CL moves the storage of data to within |_timestampMap|,
instead of being a pointer so that it's guarded by the same
lock.
Bug: webrtc:11229
Change-Id: I3f2afb568ed724db5719d508a73de402c4531dec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209361
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33374}
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}
The low-latency renderer is activated by the RTP header extension
playout-delay if the min value is set to 0 and the max value is
set to something greater than 0.
According to the specification of the playout-delay header
extension it doesn't have to be set for every frame but only if
it is changed. The bug that this CL fixes occured if a playout
delay had been set previously but some frames without any specified
playout-delay were received. In this case max composition delay
would not be set and the low-latency renderer algorithm would be
disabled for the rest of the session.
Bug: chromium:1138888
Change-Id: I12d10715fd5ec29f6ee78296ddfe975d7edab8a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208581
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33330}
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}
Before this CL, WebRTC created a decoder for each negotiated codec
profile. This quickly consumed all available HW decoder resources
on some platforms. This CL adds a field trial,
WebRTC-PreStreamDecoders, that makes it possible to set how many
decoders that should be created up front, from 0 to ALL. If the
field trial is set to 1, we only create a decoder for the
preferred codec. The other decoders are only created when they are
needed (i.e., if we receive the corresponding payload type).
Bug: webrtc:12462
Change-Id: I087571b540f6796d32d34923f9c7f8e89b0959c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208284
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33300}
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}
The SpatialIndex value from an EncodedImage is 0-based, but values were
off by 1.
Bug: none
Change-Id: Ie74e6450ddef1cfaee68fa230c441030fa86a64a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203525
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33067}
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}
by updating flag that T1 frame can be referenced when it is encoded
rather than when it is sent for encoding.
Otherwise when encoder drops T1 frame, configuration for following T2 frame would
still try to reference that absent T1 frame leading to invalid references.
Bug: None
Change-Id: I6398275971596b0618bcf9c926f0282f74120976
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202030
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33002}
This is a reland of 69241a93fb
Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
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/master@{#33001}
This reverts commit 69241a93fb.
Reason for revert: Breaks WebRTC roll into Chromium.
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
TBR=mbonadei@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
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}
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.
This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).
The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
break a circular dependency (is has been extracted from
//rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
break another circular dependency.
Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
This is just general cleanup.
The assumed behavior is late decoding, and this function is not used to make any decision (except in the deprecated jitter buffer).
Bug: webrtc:12271
Change-Id: Ifb48186d55903f068f25e44c5f73e7a724f6f456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200804
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32940}
VideoCodecInitializer::VideoEncoderConfigToVideoCodec is modified to always set correct frame rate, width and height on spatial layer 0 so the rest of the code does not need to differentiate between scalable/none scalable codecs.
Bug: webrtc:12000
Change-Id: I5a068b98ca2038621205f55e4024f949ab51587a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198540
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32890}
VideoReceiveStream2 unnecessarily posts a decode complete call to
its own queue while already being executed on it. A popular use
case in downstream project has a large amount of decoders
in use so the cost of this is multiplied by the number of active
decoders.
Fix this by removing the unnecessary task post. To allow for this,
this change also changes the frame buffer to call out to it's
handler without any locks held.
Bug: webrtc:12297
Change-Id: Ib2e26445458228a44c53dad9d585d4025f2f2945
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197420
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32845}
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}
Now that RtpVp9RefFinder sets an additional reference on the frame instead of marking it as inter_layer_predicted it is no longer used.
Bug: webrtc:12206
Change-Id: I10e0930336eafc32dc86feb2f690cb131e55be2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196740
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32814}
Instead of signaling an inter layer dependency with the inter_layer_prediction flag we instead flatten the frame IDs so that an inter layer dependency can be signaled as a regular frame reference.
Bug: webrtc:12206, webrtc:12221
Change-Id: I0390fd3d0f5494cde59eece227db938dbc5d7992
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196648
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32808}
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}
This CL breaks out descriptor specific parts into separate classes. All logic in the newly added classes is just copy pasted from the (previously massive) RtpFrameReferenceFinder with the exception of how frames are being returned, which is now done via return value rather than a callback. Basically, all interesting changes have been made in the RtpFrameReferenceFinder.
Bug: webrtc:12221
Change-Id: I5f958d2fbf4b77ba11c3c6c01d8d0d80e325be60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195448
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32717}
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}
When spatial scalability is used, both vpx and aom set key frame flag
for all spatial layers of the first frame, while rtp code expect it to
be set only on the frame without spatial dependencies.
That creates confusion for the frame dependency calculator.
Simplest solution seems to ignore that confusing signal and instead
rely encoder wrappers update frame buffer usages when key frame is generated.
Bug: webrtc:11999
Change-Id: Ica24f1d8d42d32dd24664beabf32ac24872cd15a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194002
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32667}
The Dependency Descriptor use unique ids for every frame, meaning spatial layer frames will all have unique ids.
Bug: webrtc:10342
Change-Id: I241a8b3959e27bd918ae7a907ab5158fe9dcd7a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194327
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32655}
Since inter_layer_predicted information is not propagated by the Dependency Descriptor this block non-VP9 super frames.
Bug: webrtc:10342
Change-Id: I90fbd368e92d168560a21ff79693f07071ea6cfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194140
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32643}
This CL does not aim at cleaning up simulcast/SVC configuration, just to make it possible to set the scalability mode for AV1. Implementing a codec agnostic SVC/simulcast API is a (big) project on its own.
Change-Id: Ia88df31eb1111713e5f8832e95c8db44f92887ca
BUG: webrtc:11607
Change-Id: Ia88df31eb1111713e5f8832e95c8db44f92887ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192541
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32631}
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}
Makes construction simpler, and allows the ts_extrapolator_ pointer
to be marked const.
Followup to https://webrtc-review.googlesource.com/c/src/+/190721
Bug: webrtc:12102
Change-Id: I2abeb960935b5470509f654a4a9d5121c8001900
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190981
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32535}
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}
Only use, in VCMTiming, protects it with its own mutex.
Bug: webrtc:12102
Change-Id: I9c09976f9d938565b3e2908eca6cfee0c4063f6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190721
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32529}
