When a display uses a scale factor (different than 1.0) the previous
cursor position is not properly cleared during a CRD connection on
ChromeOS (see b/235191365).
The issue was that the fix for crbug.com/1323241 does not take device
scaling into account, so that fix would incorrectly not mark the
previous location of the mouse cursor as modified.
Adding proper boundary checks is hard and risky though, as the way the
position of the mouse cursor is reported seems to be platform dependent
(ChromeOS vs Linux vs ...).
So because crbug.com/1323241 only solves a theoretical crash that is rarely if
ever hit in the field, I decided to for now undo the fix for crbug.com/1323241.
A proper boundary check can then later be introduced without any pressure from
a looming release
Bug: chromium:1323241
Bug: b/235191365
Fixed: b/235191365
Test: Manually deployed
Change-Id: Ib09b6cc5e396bd52538332edfc4395ed80c6786e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265391
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Joe Downing <joedow@google.com>
Commit-Queue: Jeroen Dhollander <jeroendh@google.com>
Cr-Commit-Position: refs/heads/main@{#37274}
Make use of "persist_mode" option in ScreenCast portal to restore
previously selected screen/window and avoid picking it again in yet
another xdg-desktop-portal dialog.
Bug: webrtc:13429
Change-Id: I3a0068091c2dd38003a7dff3f82b9cdb2ccd0f42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263901
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37257}
422 10 bit format is called I210 in the code and implemented in I210Buffer, and 420 10-bit format format is using is using the already existing I010 format and implemented in I010Buffer.
Bug: webrtc:13826
Change-Id: I6b6ed65b9fbb295386ea20f751bd0badc49ef21b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256964
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37252}
increasing precision since summing up rounded values leads to
a rounding error, in particular for small frames which take very
little time to decode.
BUG=webrtc:12526,webrtc:13756
Change-Id: I647c702808856a002c746ed9f115aa9bcaddc1f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262810
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37249}
This reverts commit 6e4d7e606c.
Reason for revert: Still breaks downstream build (though in a different way this time)
Original change's description:
> Reland "Delete old Android ADM."
>
> This is a reland of commit 4ec3e9c988
>
> Original change's description:
> > Delete old Android ADM.
> >
> > The schedule move Android ADM code to sdk directory have been around
> > for several years, but the old code still not delete.
> >
> > Bug: webrtc:7452
> > Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37174}
>
> Bug: webrtc:7452
> Change-Id: Icabad23e72c8258a854b7809a93811161517266c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265872
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37236}
Bug: webrtc:7452
Change-Id: Ide8fbd55fadd7aed9989053afff7c63c04f1320f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266023
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37242}
This is a reland of commit 4ec3e9c988
Original change's description:
> Delete old Android ADM.
>
> The schedule move Android ADM code to sdk directory have been around
> for several years, but the old code still not delete.
>
> Bug: webrtc:7452
> Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37174}
Bug: webrtc:7452
Change-Id: Icabad23e72c8258a854b7809a93811161517266c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265872
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37236}
Replace `is_chromecast` with `is_castos` and `is_cast_android` as
appropriate. See linked bug for further context.
Bug: chromium:1219802
Change-Id: If24af59e058940b7259cf4f1d9a3ba2ee0449cdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265601
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: David Dorwin <ddorwin@google.com>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Ryan Keane <rwkeane@google.com>
Cr-Commit-Position: refs/heads/main@{#37230}
This is the first CL out of three to make the low-latency stream signaling
explicit. At the moment this is done by setting the render time to 0.
There's a dependency between Chromium and WebRTC which is why this is
split into three CLs to not break any existing functionality.
Bug: chromium:1327251
Change-Id: Ie6b268746d587a99334485db77181fb2c6e9b567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264502
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37225}
When DMA-BUFs are used, sometimes stride we get from PipeWire might
contain additional padding, but after we import the buffer, the stride
we used is no longer relevant and we should just calculate it based on
width.
Bug: chromium:1333304
Change-Id: Id4300550f0b3c539ddd749e9285f525d4f816b80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265384
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37195}
This reverts commit 4ec3e9c988.
Reason for revert: Causes downstream build error.
Original change's description:
> Delete old Android ADM.
>
> The schedule move Android ADM code to sdk directory have been around
> for several years, but the old code still not delete.
>
> Bug: webrtc:7452
> Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37174}
Bug: webrtc:7452
Change-Id: If094e0a3ef5a3d340cbd5dfa0a8a88c3e97ba0bf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265393
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37180}
The method can be used to ensure packets reported to NetworkStateEstimator include transport overhead.
Change-Id: I30f0271aac82633893660c61ea59e3b7c2cf9f31
Bug: webrtc:10742
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265405
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37179}
A recent cleanup cl (r36900) had an unintended side-effect.
If the queue-time limit is expected to be hit, we adjust the pacing
bitrate up to make sure all packets are sent within the nominal time
frame.
However after that change we stopped adjusting the pacing rate back to
normal levels when queue clears - at least not until the next BWE
update (which is fairly often - but not immediate).
This CL fixes that, and also makes sure whe properly update the
adjusted media rate on enqueu, dequeue and set rate calls.
Bug: webrtc:10809
Change-Id: If00dc35169f1a1347fea6eb44fdb2868282ed3b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265387
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37178}
The schedule move Android ADM code to sdk directory have been around
for several years, but the old code still not delete.
Bug: webrtc:7452
Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37174}
It hasn't been used in years.
Bug: chromium:1331345
Change-Id: I8fdc1952fa1114f7f78e2535ffb76e9678e53d1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265520
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37164}
sprintf is marked as deprecated with Xcode 14.
Bug: chromium:1331345
Change-Id: I834f392bee96e6b6725d5aee469a243dbc6e272e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265521
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37162}
Bug: chromium:1330019
Change-Id: I1a22967dff3231c1522fb94de38b309f441d468e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265442
Reviewed-by: Frank Barchard <fbarchard@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Joe Downing <joedow@google.com>
Cr-Commit-Position: refs/heads/main@{#37158}
The same information can be found in `AudioFrame.packet_infos_`.
Bug: none
Change-Id: Ib63bc41ffb896677a445d875afce0a98acea6999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265161
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37153}
This reverts commit 0ba10283fb.
Reason for revert: This workaround is no longer needed, as the libyuv team has already fixed the underlying issue (in b/234824290)
Original change's description:
> Fix memory corruption in BasicDesktopFrame::CopyTo
>
> This memory corruption happens inside libyuv::CopyPlane()
> on platforms that support AVX. I opened b/234824290 so the libyuv team
> can investigate and fix this, but in the mean time we need to get this
> fixed asap as this is causing crashes on both M102 (which is released to
> stable) and M103 (which has this issue marked as beta blocking).
>
> Fixed: b/234824290
> Fixed: chromium:1330019
> Test: Manually reproduced on zork board
> Change-Id: I6bfd1e089020dfb23d974d3912d45c01a4e5ce26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265041
> Auto-Submit: Jeroen Dhollander <jeroendh@google.com>
> Commit-Queue: Alexander Cooper <alcooper@chromium.org>
> Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> Cr-Commit-Position: refs/heads/main@{#37121}
Fixed: b/234824290
Fixed: chromium:1330019
Change-Id: Iafc0eac651fbc7a7fce5092306b12c4377248839
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265165
Auto-Submit: Jeroen Dhollander <jeroendh@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Frank Barchard <fbarchard@google.com>
Cr-Commit-Position: refs/heads/main@{#37142}
This CL introduces PacketQueue::SizeInPacketsPerRtpPacketMediaType
keeping track of the number of packets in the queue per
RtpPacketMediaType.
The TaskQueuePacedSender is updated not to apply slack if the queue
contains any kRetransmission or kAudio packets. The hope is that not
slacking retransmissions will make the NACK/retransmission regression
of the SlackedPacer experiment go away. Wanting to not slack audio
packets is unrelated to the regression but a sensible thing to due
since audio is highest priority.
This CL does not change anything when the SlackedPacer experiment is
not running, since if its not running then none of the packets are
slacked.
Bug: webrtc:14161
Change-Id: I1e588599b6b64ebfd7d890706b6afd0b84fd746d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265160
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37139}
This memory corruption happens inside libyuv::CopyPlane()
on platforms that support AVX. I opened b/234824290 so the libyuv team
can investigate and fix this, but in the mean time we need to get this
fixed asap as this is causing crashes on both M102 (which is released to
stable) and M103 (which has this issue marked as beta blocking).
Fixed: b/234824290
Fixed: chromium:1330019
Test: Manually reproduced on zork board
Change-Id: I6bfd1e089020dfb23d974d3912d45c01a4e5ce26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265041
Auto-Submit: Jeroen Dhollander <jeroendh@google.com>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37121}
When the slacked pacer experiment is enabled the next pacing opportunity
may be a full tick (~16 ms) from now. Add a flag to allow experimenting
with a burst interval (= 16 ms?) such that we can send bursts in
MaybeProcessPackets.
A common use case would be that EnqueuePackets triggers
MaybeProcessPackets when we are off-tick but we'd still like to create
an immediate burst instead of waiting for the next tick or two for that
to happen.
Bug: webrtc:14152
Change-Id: Ib0ed8312cb7d53b80f3520fff3a6e3bbb5a93fd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264985
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37116}
Allows the PacerController to send packets in bursts. If there are enqued packets, or a packet is enqueued while the pacer have a small media debt, an enqued packet is allowed to be sent immediately as long as the debt is smaller than the set burst interval.
Bug: b/233850913
Change-Id: Ibb0fa63c97409ca23b9fa7148b5ff6ce8c4517e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264462
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37098}
Move warning about missing receive_statistics to AddReceiver to avoid
producing it for rtp send only endpoints.
Remove warning about missing cname as unimportant.
Bug: webrtc:8239
Change-Id: I8a90aa4b378284b9c68f67678b2392b9461c95b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264825
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37093}
BaseTime represents fixed point in time with unknown epoch and thus
make sense to convert to Timestamp type, however Timestamp should always
be positive. however legacy tests expect GetBaseTimeUs to return negative time sometimes.
Bug: webrtc:13757
Change-Id: I3f780a7775fdd1e271402c59384c1298db76f75a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264549
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37076}
When target_os is set to "fuchsia":
BUILD: suppress Wundef flag
DEPS: download the Fuchsia SDK
audio_encoding: add header include
video_capture: video_capture_factory is not yet implemented for Fuchsia
so we add a null capture factory when building for Fuchsia.
Bug: webrtc:14061
Change-Id: Id6ca7418859c85293a0a5e2a8427807ee039db2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262200
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37030}
Ensure each test create its own PacerController.
Move (most) operations on the pacer controller to the actual test. (the
rest should be moved too eventually....)
Use only one test fixture.
Bug: none
Change-Id: I0b8eee9d2c2f91f7102858a1a544e45e8b0b7b5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264120
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37017}
This change adds support for dynamic resolution adjustment
of pipewire stream.
Bug: chromium:1291247
Change-Id: I87e02484920f795a053a814eb872834ab22c1bd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263680
Commit-Queue: Salman Malik <salmanmalik@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37010}
u8"" no longer produces a char*. Use "" instead, which also accepts
UTF-8 literals.
Bug: chromium:1284275
Change-Id: Ida84b82670eb1238a606d3fe8c4eb40fbc23165e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263760
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37005}
It replaces the relative arrival delay tracker which is equivalent.
This results in a slight bit-exactness change but nothing that should affect quality.
Bug: webrtc:13322
Change-Id: I6ed5d6fdfa724859122928a8838acce27ac2e5d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263380
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37004}
ABSL_CONST_INIT must be on definitions, not just declarations.
Bug: chromium:1284275
Change-Id: If57064ab9417df38f770c59e50be93a104748b72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263282
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36994}
This is a continuation of https://webrtc-review.googlesource.com/c/src/+/263202
which added logging for max delay. However, if the max delay was already
set and a new min delay was set this logging could have been missed.
Bug: None
Change-Id: I2e7e5bdf920fa68aa723ec8480d564b322813712
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263480
Reviewed-by: Johannes Kron <kron@google.com>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36988}
Puts the whole block in contiguous memory and reduce pointer look-up.
The change has been verified to be bit-exact.
Bug: webrtc:14089
Change-Id: I264aaf764bf53a29f23249105f704b2fdbd7e51c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263203
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36983}
This CL allows the users to propose custom resolution to server
for the captured pipewire streams.
Bug: chromium:1291247
Change-Id: Iaae2c73df1a5f5ebac651ce7d087af4c273113c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263360
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#36979}
There are two cases that can be confusing for applications developers
which may result in the playout delay not being set as intended.
First, it is not well defined which min playout delay should be used
when multiple are set. This changes adds a warning to alert application
developers that they are setting multiple playout delays.
Second, if the playout delay header extension is used, developers must
be careful that the max playout delay is always larger than the min
playout delay, otherwise the behaviour is undefined. This change logs an
error when this case is detected.
Bug: None
Change-Id: I8477d48ef64636da080792362fa898e42f038bef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263202
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36977}
Putting these classes in a sub folder increases
structure and clarifies that they are used as
helper classes. Affected classes in this change:
* CodecTimer
* InterFrameDelay
* RttFilter
VCMTiming will be moved in a separate CL.
Additional changes:
* Remove VCM prefix from class names.
* Introduce granular BUILD.gn targets.
* Update some includes.
Bug: webrtc:14111
Change-Id: Ia75128aa955a819033b97d4784cb61904de7230b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262960
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36975}
This helper class currently lives in `modules/video_coding`,
but it's only users are in `video/`. Thus, it makes sense to
move the class to `video/`.
Bug: webrtc:14116
Change-Id: I0d3f8961bc8f5fe80f3100dbbd309b206020e6d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262963
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36973}
The high-band gain is corrected by fixing the computation of the
low-band energy
Bug: webrtc:14108
Change-Id: I5033287de57aedcd91bb71623ca2862519ffb35b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263201
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36972}
This change adds a Block class to reduce the need for std::vector<std::vector<std::vector<float>>>. This make the code
easier to read and less error prone.
It also enables future changes to the underlying data structure of a
block. For instance, the data of all bands and channels could be stored
in a single vector.
The change has been verified to be bit-exact.
Bug: webrtc:14089
Change-Id: Ied9a78124c0bbafe0e912017aef91f7c311de2ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262252
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36968}
Step one in making it a separate type, that will be done as a
followup, after downstream code is updated to use the new name.
Bug: webrtc:11607
Change-Id: I6fa664a0729b1cfd71b7f02b6441880beee0e741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262806
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36946}
AcmReceiver basically only does resampling, which is not something we need to test for bit-exactness.
NetEq bit-exactness is already tested with the same rtp input file as these tests.
Bug: None
Change-Id: Ibb3936c86098e0eea944860d33e2c13bf046e40b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262816
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36944}
to make tests faster and more determenistic.
Bug: webrtc:8239
Change-Id: I18067251a1f1a349fda28bbfbb59bce333bfddca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201737
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36938}
To make it usable in tests without depending on all of CallTest.
Bug: None
Change-Id: Ie3102ab71bcfe3862dd6c35d3285098e961e54df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262807
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36932}
Structs with user-declared constructors are not aggregates and cannot
be initialized with designated initializers. Remove declarations that
don't actually affect anything.
Bug: chromium:1284275
Change-Id: Ib45ea334d7be28bfa7bbce132985612f0e6ecd10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262820
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36916}
This is to more accurately simulate Opus CNG.
Bug: None
Change-Id: I3244d88e1f7410190551b6fa24cdd08599b5771e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262661
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36913}
Instead of using field trials in BitrateProber for probe duration, use values provided in ProbeClusterConfig from GoogCC.
Field trials are instead read in ProbeController.
To avoid having to do a thread jump for every ProbeClusterConfig, RtpPacketPacer interface is changed to RtpPacketPacer::CreateProbeClusters(std::vector<ProbeClusterConfig>
Deprecates field trial "WebRTC-Bwe-ProbingConfiguration"
Change-Id: I3991e4b54770601855a3af2d6a16678f11d41c31
Bug: webrtc:14027
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261265
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36911}
RTCPReceiver::main_ssrc_ and local_media_ssrc() represent the same
value but could get out of sync when `set_media_ssrc()` was called.
Instead of using main_ssrc_, just use the local_media_ssrc() accessor.
Bug: webrtc:11993
Change-Id: I2b034287e6b6025d9b0d2affa391a168896a614b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262663
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36905}
PipeWire server in older versions would mark the negotiation as
finished and start creating buffers.
Upstream bug: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1732
Bug: webrtc:13429
Change-Id: I7194e6672716d7fef1c2aadc40d3acf55cb282a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262621
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36901}
This completes the removal of the legacy pacer.
Bug: webrtc:10809
Change-Id: I8962ad56aa673f46b2c0e2cf8a5630e2c9942c92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262421
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36900}
A packet arrival history is used to store the timing of incoming packets and tracks the earliest and latest packets by taking the difference between rtp timestamp and arrival time. The history is windowed to 2 seconds by default. The packet arrival history will replace the relative arrival delay tracker in a follow up cl.
The playout delay is estimated by taking the difference between the current playout timestamp and the earliest packet arrival in the history. This method works better when DTX is used compared to the buffer level filter that it replaces.
The threshold for acceleration is changed to be the maximum of the target delay and the maximum packet arrival delay in the history. This prevents any acceleration immediately after an underrun and gives some time to adapt the target delay to new network conditions.
The logic when to decode the next packet after a packet loss is also changed to do concealment for the full loss duration unless the delay is too high.
The new mode is default disabled and can be enabled using a field trial.
Bug: webrtc:13322,webrtc:13966
Change-Id: Idfa0020584591261475b9ca350cc7c6531de9911
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259820
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36899}
VP9 automaticResizeOn is disabled if more than one spatial layer is configured via scalability mode.
Bug: webrtc:13960
Change-Id: I7c6351bca6d2f32bcc7391894e8dcc9e74ca2050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261315
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36897}
The new TaskQueuePacedSender has been default-on in code since M97, and
there are no further usages of it that I can find. Let's clean this up!
The PacingController and associated tests will be cleaned up in a
follow-up cl.
Bug: webrtc:10809
Change-Id: I0cb888602939add953415977ee79ff0b3878fea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258025
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36890}
This is for cleaning loss based bwe v2 implementation according to some comments from https://webrtc-review.googlesource.com/c/src/+/261240.
Bug: webrtc:12707
Change-Id: I2cb278f136cddcd0eeb2c5e4c319a9cc6aab94a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262251
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36875}
Previous version uses OSMemoryBarrier() on Apple but this is deprecated
in macOS 10.12.
Also, modify PRESUBMIT.py so that this patch is not blocked by
CheckLongLines.
Bug: chromium:1322548
Change-Id: I02c6b7682730abf718e88fc7f1bd4dcd7d347da6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256580
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36873}
HandleXEvent() returns true to indicate the event is consumed and
should not be passed to other registered handlers of the same
event-type.
In ScreenCapturerX11, this makes sense for XDamage events because they
are scoped to the object's |damage_handle_|. But RRScreenChangeNotify
and ConfigureNotify events are scoped to the root window, so this CL
changes the return value to false for these events. This allows other
handlers (including other screen-capturer instances) to see these
events.
Bug: webrtc:14060
Change-Id: Id18917b0b62d125da08578e08df9648062500cad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262142
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Lambros Lambrou <lambroslambrou@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36858}
This allows builds on non-x86 architectures such
as ppc64el.
Bug: webrtc:14057
Signed-off-by: Timothy Pearson <tpearson@raptorcs.com>
Change-Id: Ie2c1023d2c1d041ba1d140f06af432ed9e9f7432
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262002
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36856}
Apple renamed CGColorSpaceCopyICCProfile to CGColorSpaceCopyICCData in
10.13. If you compile code with a minimum OS required of 10.13 or
newer, [-Werror,-Wdeprecated-declarations] will cause an error to
occur on use of the function with the old name.
Add a compile switch so that no error is emitted regardless of the
deployment configuration.
Bug: chromium:1322548
Change-Id: Ie969aa9e5c4fc9bee2ec88b126d4c07701c3e9e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261953
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Avi Drissman <avi@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36855}
- Filter out very old packets (to ensure that the estimate doesn't drop to zero if sending is paused and later resumed).
- Discard packets older than previously discarded packets (to avoid the estimate dropping after deep reordering.)
- Add tests cases for high loss, deep reordering and paused/resumed streams to unittest.
- Remove some field trial settings that have very minor effect and rename some of the others.
- Change analyzer.cc to only draw data points if the estimators have valid estimates.
Bug: webrtc:13402
Change-Id: I47ead8aa4454cced5134d10895ca061d2c3e32f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236347
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36849}
This is in preparation for adding a portal / pipewire backend.
This just renames one class and moved the code to different files.
There are no changes to the implementation.
Bug: webrtc:13177
Change-Id: Iae101fcabafdb6cddd4d82adbb26219e4b37557f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261680
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36848}
The change bounds the estimate increment by MaxIncreaseFactor in DelayedIncreaseWindow after seeing loss. MaxIncreaseFactor is set to 1000 to disable the change by default.
Improve trendline integration: always allow to decrease the estimate, and only allow to increase the estimate if overusing and underusing are not in the state window.
Other improvement: bound candidates by delay based estimate, instance upper bound, and bandwidth limit in the current window.
Clean: remove the flag BackoffWhenOverusing since it has negative impacts when experimenting.
Bug: webrtc:12707
Change-Id: Ia4c1e58d692071967e8807a8b9d64b8ae4caf837
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261240
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36847}
Clean up by removing unused field-trial that was added in this CL
https://webrtc-review.googlesource.com/c/src/+/151911
to make it possible to simulate a slow decoder.
Bug: None
Change-Id: I237f3ac6baae76f81fcd2938e43eab9c19cea45f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261681
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36824}
Intended to let Vp8TemporalLayersFactory (an api/ target) reuse
this function, without depending on the codec implementation, and
without introducing a dependency cycle with the webrtc_vp8 build
target.
Bug: webrtc:11607
Change-Id: I671422e994e1005da8c7d768e8dd8ff795553e51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261308
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36816}
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.
Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
If feeding an odd length payload to the G722 stereo decoder, the codec
would end up reading from uninitialized memory.
Bug: chromium:1302494
Change-Id: I2222377530fee31555e17a0c60ecf33261364b71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261303
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36810}
This crash happened when:
* The cursor was located in the corner
* The screen was resized so that the cursor position is outside of the frame.
This caused us to add out-of-bound coordinates to the frame's updated_region, which caused crashes further down the pipeline.
Bug: chromium:1323241
Test: new unittest
Test: manually reproduced crash
Change-Id: Ie71db58c8a347f00af8a3803fcd55cdcad6eafac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261263
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jeroen Dhollander <jeroendh@google.com>
Cr-Commit-Position: refs/heads/main@{#36809}
Resolving https://bugs.chromium.org/p/webrtc/issues/detail?id=14023
At the moment, in DelayBasedBwe the time deltas are rounded to the
nearest millisecond. This change makes sure the numbers are passed as
doubles as expected by the TrendlineEstimator.
Change-Id: I68882547fb19af0e67e7b5d8de4159083a54d7eb
Bug: webrtc:14023
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261320
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36806}
Several tests starting failing when run on trybots on Win10. This CL
fixes several issues that were uncovered.
Issue 1:
Capture failed to start because `get_Size` returned {0, 0}. This is a
known issue in the WGC API that occurs when there are multiple user
sessions on the same machine.
Solution:
Add a `GetSize` method to the `WgcCaptureSource` interface so we can
fallback to other methods if `get_Size` fails.
Issue 2:
The screen capture tests assume there will be displays attached and
fail if there aren't.
Solution:
Always run `IsWgcSupported` for the appropriate capture type.
Issue 3:
ASAN container-overflow in `GetTestWindowIdFromSourceList`
Solution:
Check the validity of the iterator before dereferencing.
Issue 4:
Occasionally, the call to `GetMessage` in the `CloseWindowMidCapture`
test would hang because there were no messages in the queue.
Solution:
Use `PeekMessage` instead which will return if there are no messages.
Bug: webrtc:14002
Change-Id: I69b2f765db87d34a41d6a1796cd5a81f4029be33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260202
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#36802}
To quote rfc6464:
The audio level for digital silence -- for a muted audio source, for
example -- MUST be represented as 127 (-127 dBov), regardless of the
dynamic range of the encoded audio format.
The behavior in webrtc is correct that digital silence is represented
with 127, but it is also possible to get a value of 127 for not quite
digitally silent audio buffer (as in, not strictly 0s).
Bug: webrtc:14029
Change-Id: I7ff8698a7e4d5c0960c667fd1cc961838e269456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261244
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36793}
We already use RTC_NO_SANITIZE("cfi-icall") for most of the code and
it looks this one can be triggered recently with pw_loop_signal_event()
call.
Bug: webrtc:13659
Change-Id: I4dbb88f32de861e05be18254640db90b0f58c5e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261300
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36787}
* Structs with user-declared constructors are no longer considered
aggregates, so remove the declarations when possible
* Types of both arguments to "==" must match to avoid "ambiguous
function call" warning
* Various types of math involving enums are deprecated, so replace with
constexprs where necessary
* ABSL_CONST_INIT must be used on definition as well as declaration
* volatile memory may no longer be read from and written to by the same
operator, so replace e.g. "n++" with "n = n + 1"
* Replace an outdated check for no_unique_address support with
__has_cpp_attribute
* std::result_of(f(x)) has been removed, replace with
std::invoke_result(f, x)
Bug: chromium:1284275
Change-Id: I77b366ab1da7eb2c1e4c825b2714417c31ee5903
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261221
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36786}
The webrtc::VideoStreamDecoderInterface was basically created as a public version of FrameBuffer2, but to hide the complexity of FrameBuffer2 it was also combined with decoding so that the public API could be reasonably simple to use. FrameBuffer3 has a simple API with a clear purpose, so its API can be exposed directly.
Bug: webrtc:14026
Change-Id: I81dc84b869e4d16c5e02feb5c876fbcede3d4a25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261181
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36781}
This queue is a more strict round robing queue, unlike the class
named RoundRobinPacketQueue. That is, we don't have the same logic to
prioritize lower-bitrate streams.
The queue time mechanism is essentially directly copied from the
previous implementation however.
Bug: webrtc:11340
Change-Id: Ie38ba8ce27c985f5f1e907cec068d6a365089bcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260562
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36737}
This will make it easier to extend testing, implement new features (e.g.
packet culling) and experiment with new variants.
Bug: webrtc:11340
Change-Id: I747f5f6cff61e11a420e43b06ffe0c4aba438c7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260116
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36670}
There is no active use of it, and the fields are enabled by default in
the uses of it.
Change-Id: Ibfdb3f1befca886cb4b2f4b2ae4d6235aafafe3d
Fixed: webrtc:13998
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256262
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36655}
This will simplify upcoming changes to the queue.
Bug: webrtc:11340
Change-Id: Id023618fdef8a8bc9fb50e477cc87e6ba20c5421
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259875
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36644}
This avoids potentially creating structures in the queue, and removes
usage of a special priority class enabling us to move priority handling
in a follow-up.
Bug: webrtc:11340
Change-Id: I2286ef6eac62e1d6dd89f6eb6035b23f27543d8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259960
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36641}
https://webrtc-review.googlesource.com/c/src/+/259457 was a
cherry-pick to M102; as such changes were not made there to keep the
merge to just what had already landed. This addresses the issues raised
on that CL.
Bug: chromium:1316478
Change-Id: I94fad0aa6fe9c67aee5a2f2158524d75b51db48e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259660
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36631}
This reverts commit fc5871cb98.
Reason for revert: I rage quit.
Original change's description:
> Reland "Send first probe packet directly instead of enqueuing it."
>
> This is a reland of commit 8088aad5ac
>
> Patchset 1 is original CL, patchset 2 contains a fix to make behavior
> exactly identical to original. It fixes a difference when the first
> small probe packet comprises the entire probe.
>
> Original change's description:
> > Send first probe packet directly instead of enqueuing it.
> >
> > This avoids potentially creating needless containers in the packet
> > queue and removes usage of the packet prio, allowing it to be moved in
> > an upcoming CL.
> >
> > Bug: webrtc:11340
> > Change-Id: Iddd9e7e4e73c97ab25a85e42bcc0094d61fd60d3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259524
> > Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36602}
>
> Bug: webrtc:11340
> Change-Id: I7d2f476a7043edd03f44a4a3de31f13c3c946a8d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259776
> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36626}
Bug: webrtc:11340
Change-Id: Ife22cff8f5fbf9b70dc24926b84caad82d2738b8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259778
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36628}
This is a reland of commit 8088aad5ac
Patchset 1 is original CL, patchset 2 contains a fix to make behavior
exactly identical to original. It fixes a difference when the first
small probe packet comprises the entire probe.
Original change's description:
> Send first probe packet directly instead of enqueuing it.
>
> This avoids potentially creating needless containers in the packet
> queue and removes usage of the packet prio, allowing it to be moved in
> an upcoming CL.
>
> Bug: webrtc:11340
> Change-Id: Iddd9e7e4e73c97ab25a85e42bcc0094d61fd60d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259524
> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36602}
Bug: webrtc:11340
Change-Id: I7d2f476a7043edd03f44a4a3de31f13c3c946a8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259776
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36626}
This reverts commit 1386fbca8b.
Reason for revert: Now another downstream test flakes :(
Original change's description:
> Reland "Send first probe packet directly instead of enqueuing it."
>
> This is a reland of commit 8088aad5ac
>
> Original change's description:
> > Send first probe packet directly instead of enqueuing it.
> >
> > This avoids potentially creating needless containers in the packet
> > queue and removes usage of the packet prio, allowing it to be moved in
> > an upcoming CL.
> >
> > Bug: webrtc:11340
> > Change-Id: Iddd9e7e4e73c97ab25a85e42bcc0094d61fd60d3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259524
> > Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36602}
>
> Bug: webrtc:11340
> Change-Id: I3e0db452961f7f5a8d00ea3283e92ec43da2f66a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259774
> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36619}
Bug: webrtc:11340
Change-Id: I31b2c6636aca5debe8e4d71546b5a3b9eb746c9d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259862
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#36620}
This is a reland of commit 8088aad5ac
Original change's description:
> Send first probe packet directly instead of enqueuing it.
>
> This avoids potentially creating needless containers in the packet
> queue and removes usage of the packet prio, allowing it to be moved in
> an upcoming CL.
>
> Bug: webrtc:11340
> Change-Id: Iddd9e7e4e73c97ab25a85e42bcc0094d61fd60d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259524
> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36602}
Bug: webrtc:11340
Change-Id: I3e0db452961f7f5a8d00ea3283e92ec43da2f66a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259774
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36619}
This reverts commit 8088aad5ac.
Reason for revert: Downstream test failure
Original change's description:
> Send first probe packet directly instead of enqueuing it.
>
> This avoids potentially creating needless containers in the packet
> queue and removes usage of the packet prio, allowing it to be moved in
> an upcoming CL.
>
> Bug: webrtc:11340
> Change-Id: Iddd9e7e4e73c97ab25a85e42bcc0094d61fd60d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259524
> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36602}
Bug: webrtc:11340
Change-Id: Ib2a4bdad3ca82108e150feefb2680b880bc316e5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259763
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36605}
This avoids potentially creating needless containers in the packet
queue and removes usage of the packet prio, allowing it to be moved in
an upcoming CL.
Bug: webrtc:11340
Change-Id: Iddd9e7e4e73c97ab25a85e42bcc0094d61fd60d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259524
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36602}
ChromeOS will use DPI (see crrev.com/c/3322917), but the
DesktopAndCursorComposer assumed pixels were used.
Test: Manually ensured it works
Bug: b/208370410
Change-Id: I5fee50d408fd204273946009e6653d4e60d1e458
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259502
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jeroen Dhollander <jeroendh@google.com>
Cr-Commit-Position: refs/heads/main@{#36597}
Some cases have been observer where the pacing controlled might have
been stuck in a send-loop. This CL adds a circuit breaker that attempts
to break that, and provide some logging for debugging.
Bug: webrtc:11340
Change-Id: Ia3d3a44c2d897260ef23160ea7122c0d0f9baa79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259521
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36592}
Currently only implemented for codec internal CNG (Opus).
Bug: webrtc:13322
Change-Id: I00622f2967f066dba64a792e26081038ae0cb0d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259200
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36590}
TaskQueue posts delayed task in milliseconds precision. If delayed tasks
have the same wakeup time in queue, we should ensure they are waked up
in FIFO order. E.g., call `PostDelayedTask(task-i, 0)` in a loop, we
expect `task-i` is waked up as enqueue order.
Co-Author: jiahe.zhang@intel.com
Bug: webrtc:13761
Change-Id: I3bc87c2d251f8dffee868a012e828fd42e783afc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251960
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36582}
While the target has a restricted visibility, since it was in rtc_base_approved
public deps, a lot of targets were able to bypass the visibility check.
So we remove the visibility restrictions and use the dependency explicitely
everywhere instead.
Bug: webrtc:8603
Change-Id: I94a03fdf7f94c54ab72081a58dd648e2cca73d17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36566}
There is a crash report from the Windows OS API
where it repro only win10, not a Win11.
Unfortunately, Microsoft can't access the dump file or
hasn't repro internally so we decided to disable the WGC
fallback use in the OS other than Win11 now.
Once the change (support WGC fallback) reaches to Microsoft Edge
and it produce crash report, Edge team will take the
dump file to the Windows OS API owner for Win10 level fix (or
bring the Win11 fix to Win10).
Bug: chromium:1312937
Change-Id: I5335e2c57076d4fab08e9c74ade599259cff10d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258821
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36558}
XDG desktop portal utils implicitly rely on few portal interfaces.
This change makes those interfaces explicit.
Bug: chromium:1291247
Change-Id: I3c300f33d04bfa8857ac9f1f9d634dbfc077e591
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258720
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#36557}
Versions of Windows before Win11 will crash when `CreateForMonitor` is
called, but the system has no attached displays. This can be avoided by
adding a check to ensure at least one display is found before we return
true in `IsMonitorValid`. Previously we would early return `true` if the
"monitor" we were checking was the `kFullDesktopScreenId`.
Bug: chromium:1316478
Change-Id: I2562fe3834db574cf3706ee1d604472ac03f9ff3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258920
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#36555}
This CL adds flag "max_queue_time", e.g:
WebRTC-SlackedTaskQueuePacedSender/Enabled:true,max_queue_time:75ms
If the PacingController's ExpectedQueueTime() is greater than or equal
to "max_queue_time" then we will use high precision delayed tasks to
ensure a smoother (less bursty) pacing of packets.
This should only get triggered in ultra high definition scenarios or
temporarily during key frames being sent. I have confirmed manually that
with the flag set to 75 ms I get low precision when not sending key
frames and temporarily get high precision for 5-15 delayed tasks during
the sending of a key frame.
Bug: webrtc:13957
Change-Id: I3c8aeba5fe18812ed0187610cd3f92a375bc6f18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258623
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36538}
Adds two metrics for stereo detection:
- An enum indicating whether the last 10 seconds contained persistent stereo content or not, logged every 10 seconds.
- An enum indicating whether any persistent stereo content at all has been detected, logged at the end of the AEC lifetime.
These metrics allow us to assess:
- What proportion of all audio is treated as stereo.
- What proportion of sessions encounter any significant stereo content. If this is unexpectedly high, the stereo detection code may need fine tuning.
Metrics are only logged for component lifetimes exceeding 5 seconds: This is to avoid brief AEC lifetimes due to internal resets etc within APM.
Corresponding Chrome CL for XML histogram declarations:
https://crrev.com/c/3579317
Bug: chromium:1295710
Change-Id: I93e2bf74588cf4bb2a8922dbfad079bccab01456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258760
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36537}
Remote desktop wrapper needs to create a barrier on when the screencast
portal is done (either succeeded or failed). This change adds an initial
enum to differentiate it from other values. CL also generalizes the
helper `PortalFailed` to `OnPortalDone` so as to account for both
failure/success scenarios.
Bug: chromium:1291247
Change-Id: I188f72533e75a88c9b30ce2bb093dae548cef7b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258540
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#36526}
The Windows 2019 Enterprise SKU is based on RS5 but is missing support
for WGC. This changes adds a function IsWgcSupported which will check
that the API is actually present, which should be more robust than just
performing a version check.
Bug: webrtc:13932
Change-Id: Ib6a2278f316b9b86671df77fd37468c79564d655
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258163
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#36506}
During temporary stereo content when the AEC3 uses a mono reference signal, the signal is downmixed by averaging instead of using only the left channel.
Additionally, temporary stereo content is flagged as an echo path change.
Tested: Modified local build: Verified stereo mode entered / left in accordance with hysteresis and timeout thresholds. Verified temporary stereo detected during temporary stereo playout. Made an aecdump and inspected content.
Bug: chromium:1295710
Change-Id: I6bd53e615dfb3ec39bc1c73275b7d6d599ac7c57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258481
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36504}
Even if playout audio is only very briefly stereo, the AEC will enter stereo processing mode. To save CPU and improve AEC performance, this CL adds a hysteresis period before treating playout as stereo.
The feature is enabled by default in the AEC3 config.
Bug: chromium:1295710
Change-Id: I29116ab2e7823e25a02aa3b66a1c619f1d966d9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258479
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36503}
If playout audio is temporarily stereo, the AEC will currently enter stereo processing mode indefinitely. To save CPU and improve AEC performance, this CL adds support for falling back to mono after a period of no stereo.
The feature is enabled by default in the AEC3 config.
Bug: chromium:1295710
Change-Id: I690b5b22f8407f950bf41f3bcaa9ca0138452157
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258421
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36502}
This CL adds a component in the TS implementation to return a delayed
version of the voice probability values observed when `Suppress()` is
called. That is needed in order to temporally align the voice
probability values to the processed audio since TS adds algorithmic
delay.
Bug: webrtc:13663
Change-Id: I5041ace3939d2ce7ba084ae703428e66f1aa06be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255860
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36496}
The values returned by `TransientSuppressor::Initialize()` and by
`TransientSuppressor::Suppress()` are never used.
Bug: webrtc:13663
Change-Id: I20b8afb5a66f49e5ebaf132acf8bcd1c4292a5f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255822
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36492}
More robust API option that allows to fully initialize TS when created.
Bug: webrtc:13663
Change-Id: I42c38612ef772eb6d0bbde49d04ea39332a0e3c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255821
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36490}
Adding a delay unit to be used in the APM Transient Suppressor (TS)
sub-module through which the observerd voice probabilities are
temporally aligned to the audio processed by TS, which introduces
algorithmic delay.
Bug: webrtc:13663
Change-Id: I2136c303914580851c742d8db89478a13b06dacb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255680
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36487}
It is now required to specify which VAD is used to compute the speech
probability passed when `TransientSuppressor::Suppress()` is called.
In this way, it is possible to adapt parameters and/or logic of a
`TransientSuppressor` implementation to the behavior of the used
VAD. This CL also adds a "no VAD" mode option, which ignores the speech
probability argument passed when `Suppress()` and always applies mild
suppression to preserve transparency.
Finally, this CL adds a field trial to choose which VAD is used by
APM for transient suppression. Wiring the RNN VAD to TS will be done
in a follow-up CL.
Bug: webrtc:13663
Change-Id: I21ed49f91875a4ee0f04db97ea87c0dbc3db7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250962
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36485}
This is a slight change in behavior that fixes a bug where all expansions are not counted due to more than 10ms available in the sync buffer, which can happen after repeated expansions.
The counter should also be updated when in muted mode.
Bug: webrtc:13322
Change-Id: I067689ee251d3d1ae990a27cdd271f718b0d6f2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257360
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36483}
The features have two safety fallbacks:
- multichannel config has a killswitch WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch
- stereo detection has a killswitch WebRTC-Aec3StereoContentDetectionKillSwitch
Both features are enabled by default in the AEC3 config.
Tested: Bitexact on a large number of aecdumps.
Bug: chromium:1295710
Change-Id: I340cdc9140dacd4ca22d0911eb9f732b6cf8b226
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258129
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36482}
Apart from making the construction more straightforward, this change allows recreating the BlockProcessor at runtime. This is used to change parameterization at runtime in an upcoming CL [1].
[1] https://webrtc-review.googlesource.com/c/src/+/258129
Tested: Bitexact on a large number of aecdumps.
Bug: chromium:1295710
Change-Id: I2e0275c5c97044cb4370042633266b193c06b960
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258100
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36473}
This CL sets speed 9 for all resolutions when two or less cores are
available, as a heuristic for a "slow" machine.
This gives a large speed bost at a relatively small quality loss.
A field-trial kill-switch is available to override this behavior.
Bug: webrtc:13888
Change-Id: I24278a45de000ad7984d0525c47d9eb6b9ab6b60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257421
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36466}
