This reverts commit 0ba10283fb3cbdf1cedea79d84e4bc3b720da6a1.
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}
