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}
This will make it easier to find stuff...
Bug: webrtc:11943
Change-Id: I4f1ae80b40b4966cb2d8db36701bbc02ac148df6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184512
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32137}
and make git blame point to a public description
NOTRY=true
BUG=webrtc:11947
Change-Id: Ic914c30243be8fd301140bc9d9489ff5869c6461
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184502
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32130}
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
Bug: webrtc:11926
Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32129}
Single use replaced with snprintf (old code also uses snprintf, but
twice, via rtc::ToString).
Bug: webrtc:6424
Change-Id: Iedb30aacb351428974067141e166cbc53fdda180
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184365
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32127}
- Fix the minor issues with the initial library implementation.
- Add unit tests to cover basic scenarios.
Bug: none
Change-Id: Ibf28b4e20f74792fce2fe11d4780fd375a4ad3a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183343
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32122}
The SwapQueue class provides efficient and thread-safe queueing of objects that
have swap capabilities. However, the current implementation does not utilize the
user-defined swap capabilites. This CL addresses that.
Bug: b/168693942
Change-Id: Id5c97c8c9cc04579b3c26c7f1dc5f8b3362126c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184361
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32118}
After implementing transceiver.stop and associated logic with regard
to stopped media sections, there might not be a transceiver for every
media section. Allow this case.
There is a test ready for submission in Chrome:
https://chromium-review.googlesource.com/c/chromium/src/+/2410407
Bug: chromium:1127625
Change-Id: I150ea5f0da4a0cbd2bf214bc659ea0df93b607de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184343
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32117}
It should already be enabled by default in libaom, but explicitly enable
it here in case that changes.
Bug: None
Change-Id: I93a1dfc92f9c02bc5ec823c326d8cf6ff163bceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184262
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32114}
This reverts commit 60c25a303fac85dccb2ccdd9e8d6d71be13b7541.
Reason for revert: Breaks downstream iOS testers.
From an offline discussion, the GN argument is not needed and only
the 3 xctest needs the behavior of that argument. So I am reverting
this one and preparing 2 CLs to properly fix.
Original change's description:
> Reland "Switch from "rtc_ios_xctest_test" to "rtc_test"."
>
> This is a reland of 7a73c772e21983857e46cb4fcedc6cfa3f42c03e
>
> The change to fix the downstream issue is just the switch from
> "test" to "rtc_test" which is a GN template that expands to
> "test".
>
> Original change's description:
> > Switch from "rtc_ios_xctest_test" to "test".
> >
> > Using the "test" GN template instead of the "ios_xctest_test" one we
> > will get iOS support for isolates via MB and GN for free, making it
> > easier to migrate the iOS recipe and fix bugs.webrtc.org/11604.
> >
> > Bug: webrtc:11881
> > Change-Id: I72b90f8494c473fa567e6296caf7a771e4caba92
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182680
> > Reviewed-by: Dirk Pranke <dpranke@google.com>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32064}
>
> Bug: webrtc:11881
> Change-Id: Ia5338859f4e893b9f19bcca6b26b8cf66d5984e8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183766
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Dirk Pranke <dpranke@google.com>
> Cr-Commit-Position: refs/heads/master@{#32075}
TBR=mbonadei@webrtc.org,dpranke@google.com,jeffyoon@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11881, webrtc:11937
Change-Id: Ie6eea6b2a8ba5c46af40b115c6db4fd0a38a25b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32112}
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}
and add a unit test
BUG=webrtc:11796
Change-Id: I8e73b22f007c15c862faad7ca881d93c14a3a46f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184160
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32104}
This avoids the pacer thread waking up at 5ms interval if a
PeerConnection is created without actually using media.
The TaskQueuePacedSender solves the problem too, this CL is mostly a
safeguard in case we still find issues when turning it on...
Can be turned off by setting field trial "WebRTC-LazyPacerStart" to
"Disabled".
Bug: webrtc:10809
Change-Id: I8501106e608eccb14487576f24bdceaf3f324d80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183982
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32101}
The class already clears the thread that's used in its dtor
and consistently uses the same thread.
Bug: webrtc:11908
Change-Id: I5ea8d00c2e59bf46c5b369be5b23cf1d8e1875c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184060
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32097}
Deallocating the async invoker is a costly operation
but it's also unnecessary and could cause us to miss signal
events.
The data_channel_transport and data_channel_transport_invoker
are (despite the name) not related, since the latter is
used to signal events on the signaling thread whereas the
former deals with the data.
Bug: webrtc:11908
Change-Id: I37b345476a6381aef5d87807877ec1e05b380137
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184062
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32096}
Since https://webrtc-review.googlesource.com/c/src/+/161447,
frame_extra_info_ is modified from both the encoder thread and the
callback thread.
Bug: None
Change-Id: Idece4d19cae6d8363428234721616f6ca6f85832
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184121
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32095}
