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}
This reverts commit d729d12454906d924d5a142deb3432e2d5fa97ae.
Reason for revert: Breaks downstream project.
Original change's description:
> dcsctp: Use stream scheduler in send queue
>
> Changing the currently embedded scheduler that was implemented using a
> revolving pointer, to the parameterized stream scheduler that is
> implemented using a "virtual finish time" approach.
>
> Also renamed StreamCallback to StreamProducer, per review comments.
>
> Bug: webrtc:5696
> Change-Id: I7719678776ddbe05b688ada1b52887e5ca2fb206
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262160
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Victor Boivie <boivie@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37170}
Bug: webrtc:5696
Change-Id: Iaf3608b52a31eb31b4ca604539edb2e8ca89399b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265389
Auto-Submit: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#37172}
This reverts commit 869c87a2b9d9f4194d77dd30dc4175a2ecf28a74.
Reason for revert: Re-landing
Original change's description:
> Revert "Make deletion of Connection objects more deterministic."
>
> This reverts commit 942cac2e9e6a205fd673dc003a051cfb3867f2e3.
>
> Reason for revert: Reverting while downstream updates are made.
>
> Original change's description:
> > Make deletion of Connection objects more deterministic.
> >
> > This changes most deletion paths of Connection objects to go through
> > the owner class of the Connection instances, Port.
> >
> > In situations where Connection objects still need to be deleted
> > asynchronously, `async = true` can be passed to
> > `Port::DestroyConnection` and get the same behavior as
> > `Connection::Destroy` formerly gave.
> >
> > The `Destroy()` method still exists for downstream compatibility, but
> > instead of deleting connection objects asynchronously, the deletion
> > now happens synchronously via the Port class.
> >
> > Bug: webrtc:13892, webrtc:13865
> > Change-Id: I07edb7bb5e5d93b33542581b4b09def548de9e12
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259826
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36676}
>
> Bug: webrtc:13892, webrtc:13865
> Change-Id: I37a15692c8201716402ba5c10f249e4d3754ce4c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260862
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36736}
Bug: webrtc:13892, webrtc:13865
Change-Id: I29da6c8899d8550c26ccecbbd0fe5f5556c80212
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260943
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37171}
Changing the currently embedded scheduler that was implemented using a
revolving pointer, to the parameterized stream scheduler that is
implemented using a "virtual finish time" approach.
Also renamed StreamCallback to StreamProducer, per review comments.
Bug: webrtc:5696
Change-Id: I7719678776ddbe05b688ada1b52887e5ca2fb206
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262160
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37170}
This adds a stream scheduler using virtual finish time (as defined in
e.g. many Fair Queuing scheduler implementations), which indicates when
a stream's next sent packet is supposed to be sent.
In the initial version, this will be used to implement a round robin
scheduler, by emulating that a stream's virtual finish time - when
scheduled - is the "one more" than all existing virtual finish times.
That will make the scheduler simply iterate between the streams in
round robin order.
The stream scheduler component is tested in isolation, and follow-up
CLs will integrate it into the send queue.
Bug: webrtc:5696
Change-Id: Iaa2c204f9b9a00517f55355cb11cfd25bb415f9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261946
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37157}
VP9 allows to increase number of spatial layers on delta frame, which
is not supported by dependency descriptor.
Thus to generate DD compatible generic header, simulator would set max
number of spatial layers, while number of active spatial layers would be
communicated with active_decode_target bitmask
Bug: webrtc:14042
Change-Id: I4da63fa7c38b0f17758a7a6243640f444470b40c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265164
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37151}
This cl/ adds a way of setting an EncoderSelector on a specific
RtpSenderInterface. This makes it possible to easily use different
EncoderSelector on different streams within the same or different PeerConnections.
The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor.
Iff a EncoderSelector is set on the RtpSender, it will take precedence
over the VideoEncoderFactory::GetEncoderSelector.
Bug: webrtc:14122
Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37150}
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}
Make HasDataToSend not mutate any state, and let the Produce method do
all state mutation and possibly indicate if there is nothing that can be
sent. This is helpful preparation for extracting the scheduling part of
the send queue to a separate component.
Bug: webrtc:5696
Change-Id: I132779e77d3ce6a41e5fcf4432140d3728d03cdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261945
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37141}
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}
Removes all remaining usage of SetType and marks the method as
deprecated.
Bug: none
Change-Id: I98dc613978ffe7ad8a4ffd951dd974d56ed92983
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265100
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37137}
SrtpTransportInterface has been deleted, but the comment is still
retained.
Bug: None
Change-Id: I5565a29bea663a396560f7458abbe902187b1338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264660
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37131}
This is the receive-side part of supporting what is frequently called
"ndata", but actually RFC8260 - "User Message Interleaving".
This CL adds a new ReassemblyStreams implementation that can assemble
I-DATA chunks and process I-FORWARD-TSN for partial reliability.
Bug: webrtc:5696
Change-Id: I3cfbea62e7b6c02fbd3f51b43ba3fb7863cf0f88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218506
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37128}
