RequestEncoderFallback, RequestEncoderSwitch and
SetVideoCodecSwitchingEnabledRequest are now all called on the
worker thread. Before, the work already happened on that thread but
WebRtcVideoChannel adapted internally when needed.
With this CL, there are thread checks to make sure that these calls are
always made the same way, we don't need the async invoker and there
are fewer calls out from the encoder thread in VideoStreamEncoder
(reducing the chance of unintentional blocking).
Bug: webrtc:11908
Change-Id: If8738bc2a708a0fefc6fe850b32655f049f30bdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184603
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32151}
This turned out to be a bit complicated, mostly
related to the tests, but here's what's changed:
* No AsyncInvoker (and avoid ClearInternal) in
WebRtcVideoSendStream (WVSS)
* The reason it was there is due to a "design leak" from
VideoSourceSinkController/VideoStreamEncoder where the former uses
locks in all methods and is unaware of a threading model. That design
affected downstream objects, pushed the need for an async hop into
WVSS and added a lock.
A suggestion was made to address this in a follow-up change, here:
https://webrtc-review.googlesource.com/c/src/+/165684
* All methods in VideoSourceSinkController are now called on a known
and checked sequence and this CL removes the lock. This also makes
checking state consistent (i.e. calling a getter twice in a row on the
same sequence, will always return the same value, avoiding race with
other threads).
* Handling of reporting state changes from the encoder queue to the
VSSC, is done by VideoStreamEncoder.
* VideoSendStreamImpl is still instantiated on the incorrect thread [1]
but has two initialization steps [2]. The second one already runs on
the right thread. Addressing that TODO [1] is something we should do
but it has side effects to consider. For the purposes of this CL
the steps relating to the encoder (setting the sink pointer) have
been moved to [2].
[1] https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/video/video_send_stream.cc;l=94
[2] https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/video/video_send_stream.cc;drc=f4a9991cce74a37d006438ec0e366313ed33162e;l=115
Bug: webrtc:11222, webrtc:11908
Change-Id: Ie46d46e3a52bbe225951b4bd580ecb8cc9cad873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184508
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32150}
RtpVideoStreamReceiver was forked to RtpVideoStreamReceiver2
recently, so the code that checks for this parameter needs to
be present in the forked location, but it wasn't.
This also enables RtpVideoStreamReceiver2TestH264.InBandSpsPps test
on MSAN, which was another already fixed bug that wasn't ported over
to the recently forked RtpVideoStreamReceiver2.
See webrtc:11595 for information about the fork.
See webrtc:11769 for information about this fmtp parameter.
See webrtc:11376 for the original MSAN issue.
Bug: webrtc:11957, webrtc:11595, webrtc:11769, webrtc:8423
Change-Id: I3734d077b2883c2f747ad35a0189b83c1915c3ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184524
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32144}
WebRTC’s Audio Video sync can go in unbounded loop and keep on increasing audio delay if audio packets stop coming in.
The issue happens, if StreamSynchronization::ComputeDelays has:
1. relative_delay_ms = some positive value which causes avg_diff_ms_ > 30ms
2. current_audio_delay_ms < current_video_delay_ms
3. audio_delay_.extra_ms > 0 and video_delay_.extra_ms = 0
To compensate for relative delay, audio_delay_.extra_ms gets incremented every time StreamSynchronization::ComputeDelays is called by RtpStreamsSynchronizer::Process(), which happens every 1sec
RtpStreamsSynchronizer::Process() will try to set the new delay to audio stream by calling syncable_audio_->SetMinimumPlayoutDelay(target_audio_delay_ms);
This ends up calling DelayManager::SetMinimumDelay and update minimum_delay_ms_
But this update has no impact on the value returned by NetEqImpl::FilteredCurrentDelayMs (as there are no audio packets flowing in, hence neteq is not running) which is called next time RtpStreamsSynchronizer::Process(), runs and tried to compute the new audio delay (audio_info→current_delay_ms)
This causes audio delay to be increased in every iteration and it grows unbounded. I guess it will stop growing above 10sec as that is hardcoded max delay in NetEQ.
To avoid this added a check to not adjust delays when no new audio stream has come in.
Bug: webrtc:11894
Change-Id: If648f9227e43c351f887d054876cb119cc1a917e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183340
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Shyam Sadhwani <shyamsadhwani@fb.com>
Cr-Commit-Position: refs/heads/master@{#32106}
Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.
Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.
Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
It is meant for Pinpoint to run only the relevant tests when running a bisection.
The Pinpoint side of this change can be found here:
https://crrev.com/c/2404161
Bug: webrtc:11084
Change-Id: I466f39816b83e2f83a3a49845c99605f4d5a857b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183763
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32082}
Also added a FrameInfo struct to propagate various meta information along side it.
Bug: webrtc:9106
Change-Id: I1feb9f94c662c367f7c6e0a50d33705fdd5346bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183880
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32079}
Setting a minimum delay can fail in some cases. It is important that the
AV sync code is aware of failures and can act accordingly to recover and
prevent sync delays that keep increasing indefinitely.
Bug: webrtc:11805
Change-Id: I0deed951dc6c6d0905536a949af875e0a6d9f7fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183360
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32062}
We can then finally delete the top-level common_types.h, and the
corresponding build target webrtc_common.
Bug: webrtc:7660
Change-Id: I1c1096541477586d90774c7a3405b9d36edec14a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182800
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32044}
This is a reland of ba8abbb630cdd9d05e22c830d0845e920762850d
This can be relanded as the queuing issues that were causing a
crash in the WebRTC roll in Chromium have been resolved. I have
added the Chromium failing targets to the CQ for this commit and
they have succeeded.
Original change's description:
> [Adaptation] Remove QualityScalerResource when disabled.
>
> Bug: webrtc:11843
> Change-Id: I2d3e40356c266f189db0242f3c7590e6d83e4456
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181369
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31924}
Bug: webrtc:11843
Change-Id: I228331293060ef996f1dd7f8e18d52b0818f526b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182080
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31996}
Resource adaptation needs refactoring for async adaptations. For now
the resource adaptation processor can work on the encoder thread, until
it is refactored to support async adaptation.
Bug: webrtc:11867
Change-Id: I9c46da356db19c0fd52748c999ccb216f2ca923b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182040
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31991}
This is in preperation for eventual multi-stream and multi-mitigation
adaptation. This logic only applied to a single stream and thus is
better fit in the VideoStreamAdapter.
Bug: webrtc:11754
Change-Id: Icc5c7920038c82b574f4b5f7efbc92698691076f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181585
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31944}
There was a small miss in https://webrtc-review.googlesource.com/c/src/+/181369
which allowed frame dropping to re-enable, and there was no test to catch this.
This case has been fixed along with a test to ensure this isn't missed in the future.
Bug: webrtc:11843
Change-Id: I201aa451d4751586c780a07dc72a7401aed78088
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31936}
Some restrictions previously were preventing adaptation up caused by the quality resource. However, it makes sense to use the same restrictions in the case of other resources. This CL removes now unneeded wire-up of reason/resource causing adaptation.
Bug: webrtc:11771
Change-Id: Iec301a59d2a41d32d23b6be340f3b5637d697e52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181580
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31926}
Instead it should be ensured that it is started with the correct config.
This removes confusion regarding a resource state. If the resource
check is stopped then the adaptations for that resource should be
removed, and there is no way to determine that if we have one method for
stop for both reconfigure and shutdown.
Bug: webrtc:11843
Change-Id: I491f2fd1f4f803a4610124c7b0026ad75ab4a9cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181368
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31913}
Deletes all webrtc usage of this member. Next step is to delete
any downstream references, and when that's done, the member can be
deleted.
Bug: None
Change-Id: I3f3a94a063dccf56468a1069653efd3809875b01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181201
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31911}
Since the degradation preference can be fetch from the thread safe
provider, this removed the need to have 2 queues.
Bug: None
Change-Id: I94cb8b7d3d2950acfe0ad0a6d46edc038ed8e286
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177523
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31898}
It was not used by any class and all future uses can use the
VideoSourceRestrictionsListener.
Bug: webrtc:11834
Change-Id: I5c71b93cc503f458dce0ccdd78b91b5a1debc56d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181062
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31896}
This way can double adapt right away instead of relying
on the qp scaler checking soon into the future.
Bug: webrtc:11830
Change-Id: I8e878168303cf6a4c3edcf3997dd8ac2413a4479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181060
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31895}
This is a reland of 32ca95145c4636374266f5b5d4d1ac43658bc758
Fix includes not enabling the screenshare conference behavior on non
screenshare sources even if the flag is enabled.
Original change's description:
> Only enable conference mode simulcast allocations with flag enabled
>
> Non-conference mode simulcast screenshares were mistakenly using the
> conference mode semantics in the simulcast rate allocator, which broke
> spec compliant usage in some situation.
>
> This behavior should only be used when explicitly using the SDP entry
> "a=x-google-flag:conference" in both offer and answer.
>
> Bug: webrtc:11310, chromium:1093819
> Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31828}
Bug: webrtc:11310
Bug: chromium:1093819
Change-Id: Ic933f93a5c4bad20583354fe821f8a1170e911cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180802
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31847}
This reverts commit 32ca95145c4636374266f5b5d4d1ac43658bc758.
Reason for revert: Internal test failure
Original change's description:
> Only enable conference mode simulcast allocations with flag enabled
>
> Non-conference mode simulcast screenshares were mistakenly using the
> conference mode semantics in the simulcast rate allocator, which broke
> spec compliant usage in some situation.
>
> This behavior should only be used when explicitly using the SDP entry
> "a=x-google-flag:conference" in both offer and answer.
>
> Bug: webrtc:11310, chromium:1093819
> Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31828}
TBR=ilnik@webrtc.org,hta@webrtc.org,orphis@webrtc.org
Change-Id: I5ccb6e87594f491ba09fe6b837ee24d63db878ca
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11310
Bug: chromium:1093819
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180801
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31829}
Non-conference mode simulcast screenshares were mistakenly using the
conference mode semantics in the simulcast rate allocator, which broke
spec compliant usage in some situation.
This behavior should only be used when explicitly using the SDP entry
"a=x-google-flag:conference" in both offer and answer.
Bug: webrtc:11310, chromium:1093819
Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31828}
RTPFragmentationHeader is already ignored by H264 packetizer
and thus doesn't need to be provided and calculated.
Bug: webrtc:6471
Change-Id: I45bc22827f0dc811457e3ebe477a16293501c2fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179843
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31791}
