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}
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}
Currently is_linux is set to true on Chrome OS build,
but it is planned to be set false. This CL is the preparation
to keep the compatibility.
Bug: chromium:1110266
Test: Build locally.
Change-Id: Ic79a202b0b3baeff157955cd03a07556bfb958a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Hidehiko Abe <hidehiko@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32073}
This method is used by GetStats, and hence must not modify any state.
This cl is step one of the api change, the non-const version of the
method can be deleted once downstream implementations of this
interface are updated.
Bug: webrtc:11622
Change-Id: Icfaccee6bc918ac5c8a39dd2567a1081e342e9e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183881
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32072}
This bypasses the proxy for the following properties:
* MediaStream::id()
* AudioTrack::kind() and AudioTrack::id()
* VideoTrack::kind() and VideoTrack::id()
* RtpReceiver::media_type() and RtpReceiver::id()
* RtpSender::media_type() and RtpSender::id()
* VideoTrackSource::remote() and VideoTrackSource::is_screencast()
* RtpTransceiver::media_type()
Bug: webrtc:11923
Change-Id: If7edea1781f778af3775515fc4af9a9e151c8103
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183767
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32071}
This CL adds explicit initialization of the FilterAnalyzer in AEC3.
While the current code never uses any fields before they are initialized,
it makes sense to be on the safe side and add initialization during
construction.
Bug: webrtc:11918
Change-Id: I467c4c8b8d6dd859a1b216baef28ac1e9d3f76c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183764
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32069}
This reverts commit 7a73c772e21983857e46cb4fcedc6cfa3f42c03e.
Reason for revert: Breaks downstream 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}
TBR=mbonadei@webrtc.org,dpranke@google.com,jeffyoon@google.com
Change-Id: Ia4d6257fee42661c10303217980bd0a9126d2709
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11881
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183765
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32068}
The signature of send_cb was changed, adding ulp_info. This change makes
it easier to retrieve the SctpTransport pointer from the callback.
Bug: webrtc:11899
Change-Id: I12a4ccd2d0deb329f6be17a4c7208449833dc188
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182984
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32067}
For Non-DPI aware windows, we need to figure out the current DPI
and scale the content accordingly, the current behavior works ok
for until the clipped region pushes the content outside of the
frame and then the capture will fail. When this happens, the
captured frame may be blank or it could cause the browser to crash.
The issue is that the left and top clipped regions are not being
scaled along with the content (the captured window region is
contained within a larger window frame). When the clipped window
and window frame are scaled, the original offset for left and top
are not adjusted so after a certain DPI, this offset causes the
clipped region to get pushed outside of the frame which is why
the capture fails.
The fix is to scale the left and top clipped regions and translate
the clipped region accordingly. This change will only affect non-DPI
aware windows.
Bug: chromium:1083527
Change-Id: I893c2cb362cbaa01170d1e58465e43c3517139ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183660
Commit-Queue: Joe Downing <joedow@google.com>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32065}
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}
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}
