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}
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}
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}
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}
All it provides is a method to call a signal on the network thread,
so it's not worth the added complexity. Implementations of
NetworkMonitorInterface must hop to the network thread anyway to
guard their members.
Also added some thread annotations to AndroidNetworkMonitor.
Bug: webrtc:9883
Change-Id: I64bb82ea593433f3a52871dbb75eb2ac4f47d69c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181420
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32087}
This needs to be followed immediately by a CL that adds unit tests for
CancerStickCastle and UntypedFunction.
Bug: none
Change-Id: I5ade68cc4721d7442db7695f218ecd9be1d639ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182460
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32085}
Currently test code passes pointer to temporary objects, while
RtcpSender passes raw pointers to objects that are then seen as owned,
and will be manually deleted by a overloaded destructor, which is scary
and fragile.
This CL moves all usage to std::unique_ptr<RtcpPacket> instead, which
may create some heap churn in unit tests but that should be fine.
Bug: webrtc:11925
Change-Id: I981bc7ccd6a74115c5a3de64b8427adbf3f16cc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183920
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32084}
