This reverts commit 6e4d7e606c4327eaa9298193e22794fcb9b30218.
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 4ec3e9c98873520b3171d40ab0426b2f05edbbd2
>
> 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 4ec3e9c98873520b3171d40ab0426b2f05edbbd2
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 4ec3e9c98873520b3171d40ab0426b2f05edbbd2.
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 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}
