All calls in code under test were migrated to AudioEncoderFactory::Create and thus there is no longer need to propagate older function.
Bug: webrtc:343086059
Change-Id: I9e0ea4024759deb22c0d284e0e4bac7322a08f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357181
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42638}
This is a cleanup of simulcast.cc. Remove GetNormalSimulcastLayers and GetScreenshareLayers from simulcast.h. Move the implementations to anonymous namespace in simulcast.cc.
Bug: webrtc:351644568, b/352504711
Change-Id: Iff03161e5c44cb0e7faa60b16cfc2fc9b903d5ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357103
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42635}
This is a cleanup of simulcast.cc. max_qp is not needed to decide simulcast config. Move setting of max QP in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams(), where it can be set per stream.
Bug: webrtc:351644568, b/352504711
Change-Id: Ia0e3e9d90032383574dc8867b30d362e9c5df7e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357102
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42634}
This is a cleanup of simulcast.cc. bitrate_priority is not needed to decide simulcast config. Move setting of bitrate priority in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams().
Bug: webrtc:351644568
Change-Id: I002d728ccf8d141fe4bbb32b390129ce57c830cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357101
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42629}
num_ref are represented using golomb and may be very large.
BitstreamReader is generally resilent to many consenquites fail reads, but not when number of reads comparable to int limit.
This change address the issue in two ways, either one is enough, but both are helpful in their own way:
H264 parser now fails faster when number of references is too large.
BitstreamReader now is resilent to unlimited number of fail reads.
Bug: chromium:352402499
Change-Id: I19646bc3f53cd2970393d00bc143400b1fdf5473
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357100
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42628}
This reverts commit 4f3d660f4f28a1dcdb577ab89d57830083ac883a.
Reason for revert: downstream still broken
Original change's description:
> Reland "Remove jni_zero type aliases in jni_generator_helper.h."
>
> This is a reland of commit 9fcaa034bc032da9de5d6fcdd45528169f44d343
>
> Original change's description:
> > Remove jni_zero type aliases in jni_generator_helper.h.
> >
> > This CL removes some type alias from
> > sdk/android/src/jni/jni_generator_helper.h and make sure all the
> > jni_zero types are referred to using the jni_zero:: namespace.
> >
> > The goal is to remove sdk/android/src/jni/jni_generator_helper.h
> > in future CLs.
> >
> > Bug: b/319078685, b/351773023
> > Change-Id: Ief60fce3e8f301f09ac5392d143aa5a82a445bcb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356882
> > Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
> > Reviewed-by: Jeremy Leconte <jleconte@google.com>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42618}
>
> Bug: b/319078685, b/351773023
> Change-Id: I604b0842d220d76c36b73d2d49bcefe0ee7ae14f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356903
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42626}
Bug: b/319078685, b/351773023
Change-Id: I9429789e44c82735305b2d0d90cbaa4891d79622
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357041
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42627}
This is a reland of commit 9fcaa034bc032da9de5d6fcdd45528169f44d343
Original change's description:
> Remove jni_zero type aliases in jni_generator_helper.h.
>
> This CL removes some type alias from
> sdk/android/src/jni/jni_generator_helper.h and make sure all the
> jni_zero types are referred to using the jni_zero:: namespace.
>
> The goal is to remove sdk/android/src/jni/jni_generator_helper.h
> in future CLs.
>
> Bug: b/319078685, b/351773023
> Change-Id: Ief60fce3e8f301f09ac5392d143aa5a82a445bcb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356882
> Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42618}
Bug: b/319078685, b/351773023
Change-Id: I604b0842d220d76c36b73d2d49bcefe0ee7ae14f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356903
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42626}
ScreenCapturerSCK uses some fields that were not available in macOS 13
but the code compiles with the older SDK because of missing annotations
that were added in the macOS 15 SDK.
Bug: chromium:351843815
Change-Id: Ic1a89b4cab43d6ee81d447ccc33ef94439752c45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356860
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Lambros Lambrou <lambroslambrou@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42624}
Call it once instead of 3 times.
Also remove FindSimulcastMaxBitrate, FindSimulcastTargetBitrate, FindSimulcastMinBitrate and a one parameter less version of InterpolateSimulcastFormat.
Bug: b/337757868, webrtc:351644568
Change-Id: I7b4002fc3134c47f368bb833925c959a07ce5177
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356980
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42623}
MakeAudioEncoder planned to be removed, and Create planned to become pure virtual
While at it, cleanup nearby mock usage:
Remove ON_CALL that by default return default constructed result
Remove EXPECT_CALL().Times(AnyNumber()) for a NiceMock
Remove parameters in EXPECT_CALL when all are wildcard
Remove redundant get to deference a smart pointer
Bug: webrtc:343086059
Change-Id: Ica90a4980350cb82bcebd11df6c63a01b828bb9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356884
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42622}
Current thresholds were tuned to guarantee no buffer overshoot in an extreme scenario (encoding a high complexity video in a low bitrate).
Bug: b/337757868, webrtc:351644568
Change-Id: I832b2564af6f18f06550338cc9b3618f8acdf831
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356580
Reviewed-by: Dan Tan <dwtan@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42620}
This reverts commit 9fcaa034bc032da9de5d6fcdd45528169f44d343.
Reason for revert: Breaks downstream project.
Original change's description:
> Remove jni_zero type aliases in jni_generator_helper.h.
>
> This CL removes some type alias from
> sdk/android/src/jni/jni_generator_helper.h and make sure all the
> jni_zero types are referred to using the jni_zero:: namespace.
>
> The goal is to remove sdk/android/src/jni/jni_generator_helper.h
> in future CLs.
>
> Bug: b/319078685, b/351773023
> Change-Id: Ief60fce3e8f301f09ac5392d143aa5a82a445bcb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356882
> Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42618}
Bug: b/319078685, b/351773023
Change-Id: I003acf68e2b84bd0a5cda7c7180a28bcd3ca3772
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356902
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42619}
This CL removes some type alias from
sdk/android/src/jni/jni_generator_helper.h and make sure all the
jni_zero types are referred to using the jni_zero:: namespace.
The goal is to remove sdk/android/src/jni/jni_generator_helper.h
in future CLs.
Bug: b/319078685, b/351773023
Change-Id: Ief60fce3e8f301f09ac5392d143aa5a82a445bcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356882
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42618}
The function has been deprecated in favor of
FrameEncodeMetadataWriter::FillMetadataAndTimingInfo().
Bug: chromium:328598314
Change-Id: Iaf2008e855dbd71f2d7cf412d95c5932b3645d71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356042
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42613}
This CL adds the sequence checks to the functions that creates or
destroys the encoder context. Those functions must be executed on
the webrtc encoder sequence.
Bug: b/320555128
Change-Id: I1daa93f2f5326073e8d75e1d711d7554bed76a62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Hirokazu Honda <hiroh@google.com>
Cr-Commit-Position: refs/heads/main@{#42612}
Similar to BGRA/RGBA we added recently, formats from PipeWire are in
big-endian, while WebRTC (using libyuv) is little-endian, therefore we
have to map BGR to RGB and not RGB to RGB as colors would be off. Also
add some additional formats supported by libyuv.
Bug: webrtc:42225999
Change-Id: Iee8303f0922fe434069b2b3f88994abecf7d2cc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355860
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42609}
* Simplified ctor. Get settings (max_qp, content_type, etc) from encoder_config passed to CreateEncoderStreams().
* Some tests assigned VideoEncoderConfig::video_stream_factory to EncoderStreamFactory they created. That's not really needed. VideoStreamEncoder creates the factory if video_stream_factory is not provided [1]. Removed video_stream_factory initialization in tests.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/video_stream_encoder.cc;l=1002;drc=1d7d0e6e2c5002815853be251ce43fe88779ac85
Bug: b/347150850, webrtc:42233936
Change-Id: Ie0322abb6c48e1a9bd10e9ed3879e3ed484fea5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355321
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42608}
Only minor positive changes seen in initial testing. Let's default-
enable and monitor behavior through the normal release cycle.
Bug: b/349561566
Change-Id: Id6b39daa159068bf076acc34888b5d7eaf110329
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356641
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42607}
For better consistency with the rest codebase (it is min_/max_ for all params in video_encoder.h; only qp is for some reason prefixed with minimum_).
Also fixed constant names in libaom AV1 encoder wrapper (moved min from suffix to prefix, minimum -> min_).
Bug: chromium:328598314
Change-Id: I6d8521a3abff3a0595a5241c02ef4746eb4694df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42604}
Before this change the AV1 encoder wrapper converted target frame rate from double to integer with rounding to the middle. That approach resulted in a bitrate mismatch caused by rounding error. The mismatch was especially high at low frame rates. For example, at target frame rate 1.4fps the bitrate mismatch reached 40%:
out/debug/video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --framerate_fps=1.4 --width=320 --height=180 --bitrate_kbps=32 --num_frames=600
...
RESULT s0t0_bitrate_mismatch_pct: DISABLED_EncodeDecode= {39.171875,0} n%
After the change the mismatch reduced to ~2% in the same scenario:
RESULT s0t0_bitrate_mismatch_pct: DISABLED_EncodeDecode= {-2.178125,0} n%
Bug: b/337757868
Change-Id: Ia51f92b3dfdce103eed1d04cac0e084b69fa8213
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356500
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42601}
There's some test code associated with this code path that can
be deleted, so this is a first step towards removing it. From what
I can tell, this is never used.
Bug: none
Change-Id: Idfb8a6c58b929c2eedd0cfc7bdc72f5b3862f5bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356481
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42600}
This is cherry-picked from WebKit's patch for fixing a fuzzer failure.
The original patch: https://github.com/WebKit/WebKit/pull/30438
Bug: chromium:41480904
Change-Id: Ic8eddb9de816c4c8d720dac6d4c55d1db3f0596e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#42598}
in addition to last received frame rtp timestamp. This helps in cases
where frames continue to be received but the decoder fails to decode.
BUG=None
Change-Id: I56ad5f9ef85cc598d3c1a1971c4c697eb6b70165
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356080
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Markus Handell <handellm@google.com>
Cr-Commit-Position: refs/heads/main@{#42596}
This can be tested by trying to connect to a TURN server that does not
listen on a specific TCP port.
BUG=None
Change-Id: I7029112afa4b1b4376220dfc2d613a30090e4f7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354901
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42595}
If traffic policing is enforced by dropping packets, RTT can still be low.
If a packet is dropped that is needed to contninue decoding, it make sense that a nack request is sent until the packet is received, or a new key frame is requested. A key frame will be requested after 3s.
For now, this cl only increase the number of times a packet can be requested.
Bug: b/317178411
Change-Id: Iea75d36ed06f346af1dd4e55a9961d5eca45f519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356482
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42594}
Behind a flag, the new behavior changes how the "media rate" utilization
is calculated:
* Instead of per spatial & temporal layer, it's per spatial layer only.
* Overshoot is compared to real target vs adjusted target.
* Window takes quite periods/frame drops more into consideration.
This should lead to less push-back when not network constrained and
complex content is used causing bursty behavior.
Bug: b/349561566
Change-Id: I402e6531183493c963fec48ae363ce0b859b396a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356480
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42593}
This class measures the allocated cumulative byte budget (as specified
by one or more rate updates) and the actual cumulative number of bytes
produced over a sliding window.
A utilization factor (produced bytes / budgeted bytes) is calculated
seen from the first data point timestamp until the last data point
timestamp plus the amount time needed to send that last data point
given no further updates to the rate.
Wireup to EncoderBitrateAdjuster will happen in a follow-up CL.
Bug: b/349561566
Change-Id: Id0dc183b07a96366531007be9ff1c1ec6574e9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356200
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42591}
This helps with making AudioBuffer compatible with current and upcoming
code that uses audio_views.h (a simpler abstraction).
Bug: chromium:335805780
Change-Id: Ib59bba274c7abfb441e3c4d606f804b365df236d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355844
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42590}
