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}
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}
This is the next step towards making MessageHandler a pure virtual
interface. All dependencies that require automatic cleanup
should be depending on the MessageHandlerAutoCleanup class.
Next step will be to remove the ctor from MessageHandler and make
it a pure virtual interface.
Bug: webrtc:11908
Change-Id: I9321b6d9e57c167868f8b896a5345fbfe19af0e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183984
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32090}
