The problem was fixed by implementing the methid PacketDuration() in
AudioDecoderG722StereoImpl, which catches the issue in
AudioDecoder::Decode().
Bug: chromium:1280851
Change-Id: I31f974b9999f3c1c62b0e5dc39bb3e56a9a9388d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251842
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36034}
On Android, MediaCodec can request a specific layout of the input buffer.
One can use the stride and slice height to calculate the layout from
the Encoder's MediaFormat. The current code assumes
a specific layout, which is a problematic in Android 12.
Fix this by honoring the stride and slice-height.
Bug: webrtc:13427
Change-Id: I2d3e429309e3add3ae668e0390460b51e6a49eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#36033}
This is purely to aid with `git log` type statements that allows for
grouping different display names for the same address.
No-try: true
Bug: none
Change-Id: I6b0af50eac356aa864e1387f3f35c3270c211faf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251941
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36032}
The flags isolated-script-test-output and isolated-script-test-perf-output need to be consumed by the tests.
The generated .app folder in added in the data list of the gni file.
This will make it available in the runtime_deps file and thus will be populated to the swarming tasks.
Bug: webrtc:13556
Change-Id: I2c75774b847d9f686c3abc00ba0400bbc3fcefae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240520
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36029}
To disable dcSCTP and fallback to usrsctp, you can use the field trial
WebRTC-DataChannel-Dcsctp/Disabled/
Also remove a hidden no-break space in dcSCTP logging causing issues in
some log parsing.
Bug: chromium:1243702
Change-Id: I46136a8913a6d803a3c63c710f3ed29523e4d773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251867
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36027}
Minor related updates to AudioTrack and VideoTrack's sequence checkers.
There's more that can be done (or arguably needs to), but this is
a start.
Bug: none
Change-Id: I3ccf8eb9bbb6bef62b83248a23a68871b9fcd9e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251843
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36021}
To disable dcSCTP and fallback to usrsctp, you can use the field trial
WebRTC-DataChannel-Dcsctp/Disabled/
Bug: chromium:1243702
Change-Id: Ia90b796562245558a61481317bcded437400b045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251800
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36018}
Current implementation has mouse cursor as part of the screen itself
which means that everytime a cursor changes location, we have to update
whole screen content, which brings unnecessary load overhead. Using our
own mouse cursor monitor implementation allows us to track only mouse
cursor changes and update them separately for much better performance.
Bug: webrtc:13429
Change-Id: I224e9145f0bc7e45eafe4490de160f2ad4c8b545
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244507
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36011}
This has mostly seemed to work fine until now; but there's a collision
happening in chromium where if the source is being shown in the Window
Picker it collides with the (also null) Dialog ID and is ignored. While
we could patch that code to not count Null as a collision, there's the
potential for other (future) code to simply ignore a capture source
that it thinks is Null.
Fixed: chromium:1295375
Change-Id: I4356084f0af97f4d56632938b0d9a24d327f7107
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251500
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36008}
This CL sets all visibility to ":*" (this buildfile) where no users
outside this directory are known, and marks up publicly exported
targets and Chrome dependencies explicitly.
Bug: webrtc:13661
Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36004}
Ensures that frames are decoded instantly when in low-latency render
mode. This also tests the max queue size behaviour. Adds a new test
suite for FrameBufferProxy that sets the appropriate field trials.
* Fixes FrameDecodeTiming to never use negative wait times for decode
timestamps.
R=kron@webrtc.org
Change-Id: I06cbec52e1e866e21aa964b24c4fd0163c26961b
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35999}
With more GPUs it might happen that server used different render
node from the one we pick from the list. This would cause DMA-BUF to
fail to import so we use Wayland client library to obtain wl_display in
order to initialize EGLDisplay using same render node and have previous
approach as a fallback. Also everyone else uses EGL_LINUX_DMA_BUF_EXT
target for importing EGLImages from DMA-BUF file descriptors so use it
as well to be sure we import buffers same way as they are produced.
Bug: chromium:1290566
Bug: webrtc:13429
Change-Id: I32bbb0bdb28c08b6e7fcb3f94009f82a2041b6ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250661
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35997}
Wires up DecodeSynchronizer in Call if there is a Metronome injected
into the PeerConnectionFactoryDependencies.
Change-Id: I362cd12648bfa0c32e73111fcd0f3296fca2b275
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251341
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35996}
This reverts commit 1158bff15f33c467543928dd6a49cb6ad04da1ba.
Reason for revert: Downstream issues unresolved (2nd of two reverts)
Original change's description:
> Reland "Use non-proxied source object in VideoTrack."
>
> This is a reland of 3eb29c12358930a60134f185cd849e0d12aa9166
>
> This reland doesn't contain the AudioTrack changes (see original
> description) that got triggered in some cases and needs to be
> addressed separately.
>
> Another change in this re-land is that instead of the `state` property
> of the VideoTrack be marshalled to the signaling thread, it's readable
> from the calling thread. Previously this was marshalled to the worker
> and the original changed that to the signaling thread (same as for
> AudioTrack) - but in case that's causing downstream problems this reland
> uses BYPASS_PROXY_CONSTMETHOD0 for the `state()` accessor of the
> VideoTrack proxy.
>
> Original change's description:
> > Use non-proxied source object in VideoTrack.
> >
> > Use the internal representation of the video source object from the
> > track. Before there were implicit thread hops due to use of the proxy.
> >
> > Also, override AudioTrack's enabled methods to enforce thread
> > expectations.
> >
> > Bug: webrtc:13540
> > Change-Id: I4bc7aca96d6fc24f31ade45e47f52599f1cc2f97
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250180
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35911}
>
> Bug: webrtc:13540
> Change-Id: Icb3e165f07240ae10730a316d3a8a3b2b9167d82
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251387
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35979}
# Not skipping CQ checks because original CL landed > 1 day ago.
Using "No-Try" to not have to wait for the win chromium bot to unblock
(currently takes hours).
No-Try: true
Bug: webrtc:13540
Change-Id: I8f34536bf472a6d069344e84d889864f195c93f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251686
Reviewed-by: Christoffer Jansson <jansson@google.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35993}
Add optional offset-to-UTC parameter to output. This allows aligning
the x-axis in the generated charts to other UTC-based logs.
Bug: b/215140373
Change-Id: I65bcd295718acbb8c94e363907c1abc458067bfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250203
Reviewed-by: Kristoffer Erlandsson <kerl@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35992}
This reverts commit f342d6054ad984b7b80df2afe349c3bbb5f1d5b8.
or "Reland "Use non-proxied source object in VideoTrack.""
Reason for revert: Didn't resolve the downstream issues.
Original change's description:
> Revert "Reland "Use non-proxied source object in VideoTrack.""
>
> This reverts commit 1158bff15f33c467543928dd6a49cb6ad04da1ba.
>
> Reason for revert: This is a partial revert as we're tracking down
> the source of the downstream issues. This CL reverts the use of
> `internal()` for methods that relate to the source sink.
>
> Original change's description:
> > Reland "Use non-proxied source object in VideoTrack."
> >
> > This is a reland of 3eb29c12358930a60134f185cd849e0d12aa9166
> >
> > This reland doesn't contain the AudioTrack changes (see original
> > description) that got triggered in some cases and needs to be
> > addressed separately.
> >
> > Another change in this re-land is that instead of the `state` property
> > of the VideoTrack be marshalled to the signaling thread, it's readable
> > from the calling thread. Previously this was marshalled to the worker
> > and the original changed that to the signaling thread (same as for
> > AudioTrack) - but in case that's causing downstream problems this reland
> > uses BYPASS_PROXY_CONSTMETHOD0 for the `state()` accessor of the
> > VideoTrack proxy.
> >
> > Original change's description:
> > > Use non-proxied source object in VideoTrack.
> > >
> > > Use the internal representation of the video source object from the
> > > track. Before there were implicit thread hops due to use of the proxy.
> > >
> > > Also, override AudioTrack's enabled methods to enforce thread
> > > expectations.
> > >
> > > Bug: webrtc:13540
> > > Change-Id: I4bc7aca96d6fc24f31ade45e47f52599f1cc2f97
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250180
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35911}
> >
> > Bug: webrtc:13540
> > Change-Id: Icb3e165f07240ae10730a316d3a8a3b2b9167d82
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251387
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35979}
>
> TBR=tommi@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I4d8e3aced019215b97a6263cafa2a7488cd118be
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:13540
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251661
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35983}
# Not skipping CQ checks because original CL landed > 1 day ago.
Using "no-try" since a follow-up revert is also needed to get the bots
to turn green.
No-try: true
Bug: webrtc:13540
Change-Id: I361fca6949c01200d9d706749e7e825eb5b4fc1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251685
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35991}
Adds new class DecodeSynchronizer that will coalesce the decoding
of received streams on the metronome. This feature is experimental and
is backed by a field trial WebRTC-FrameBuffer3.
This experiment now has 3 arms to it,
"WebRTC-FrameBuffer3/arm:FrameBuffer2/": Default, uses old frame buffer.
"WebRTC-FrameBuffer3/arm:FrameBuffer3/": Uses new frame buffer.
"WebRTC-FrameBuffer3/arm:SyncDecoding/": Uses new frame buffer with
frame scheduled on the metronome.
The SyncDecoding arm will not work until it is wired up in the follow-up
CL.
This change also makes the following modifications,
* Adds FakeMetronome utilities for tests using a metronome.
* Makes FrameDecodeScheduler an interface. The default implementation is
TaskQueueFrameDecodeScheduler.
* FrameDecodeScheduler now has a Stop() method, which must be called
before destruction.
TBR=philipel@webrtc.org
Change-Id: I58a306bb883604b0be3eb2a04b3d07dbdf185c71
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250665
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <holmer@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35988}
