This reverts commit 626f87d90501fd8d7a4ea071686cd8befd0d430c.
Reason for revert: Breaks one downstream project, will re-land after the dependency stops referencing an unimplemented RTT metric
Original change's description:
> [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
>
> In preparation for the spec moving closer to PR, let's not have
> placeholder metrics not implemented.
>
> Bug: webrtc:14167
> Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37205}
Bug: webrtc:14167
Change-Id: I7e9ac60eb474b44fab678d4c08ddcae846ce456c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265800
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37206}
In preparation for the spec moving closer to PR, let's not have
placeholder metrics not implemented.
Bug: webrtc:14167
Change-Id: If4688ef85b57f88154d490186b306b30414874e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37205}
In FrameBuffer3Proxy, if the stream became undecodable for a long
period of time and during this period the FPS changed,
the render times and decode delays would stray and cause
video pauses. This was because FrameBuffer3Proxy only updated the rtp
timestamp extrapolator on each new decodable temporal unit, rather than
each new frame.
Bug: webrtc:14168
Change-Id: I67a2c9ea392d24f84e82aa04f8c3076de11732af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265388
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37201}
When a large queue of frames builds up due to a lost frame, the decode
delay can sometimes become quite large. In this case the stream may
signal as timed out when in fact it is not. Instead, the delay should
be capped at the timeout limit.
Bug: webrtc:14168
Change-Id: I5b4e8851b2c6d7d27a698627dc1633931d7fc00e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265404
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37199}
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}
RFC 4960, 7.2.4. Fast Retransmit on Gap Reports
4) Restart the T3-rtx timer only if the last SACK acknowledged the
lowest outstanding TSN number sent to that address, or the
endpoint is retransmitting the first outstanding DATA chunk sent
to that address.
The second part of this sentence, "or the endpoint is retransmitting the
first outstanding DATA chunk sent to that address."
This means that on fast retransmit, and if the retransmitted chunks
weren't quickly acknowledged by the peer, the retransmission timer would
expire too quickly. With this CL, it will restart, as it should.
Bug: webrtc:12943
Change-Id: I7627253718530876acc516460562e66bfcc79533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37193}
Baremetal machines have webcams and it makes it confusing wether webrtc_perf_tests require cameras to run.
Change-Id: I1f3dc05c976ed008079f990b9a55f3310ea73dda
Bug: b/235780120
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265641
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#37189}
I've added a basic AV1 impl to Chrome Remote Desktop and am looking into
what is needed to test with I444 (Profile-1) in our platform. This CL
adds a few helper functions, constants, and enums that can be used to
configure the SDP with different AV1 profiles. More work is still needed
but I wanted to get this in place first so I can build on it in the CRD
host code.
Change-Id: I1af9ebf31f833138e8c36e0c0a30e32289e7b58e
Bug: chromium:1329660
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264000
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Joe Downing <joedow@google.com>
Cr-Commit-Position: refs/heads/main@{#37182}
Extract NamesCollection into separate file and add ability to
remove peer after it was added. In such case we need to preserve
old indexes, because DVQA still may request removed name from
collection due to async processing.
Bug: b/231397778
Change-Id: I87bdfb4653e7eca50d311482553d2353b1d9974e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265394
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37181}
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}
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}
