When ScreencastPortal::OnStartRequestResponseSignal receives either a
non-zero response code or is missing the response data, it would
directly cast this to a RequestResponse. However, this direct cast is an
error. Per the documentation, the response signal returns the following
values with their corresponding meanings:
0 - Success
1 - User Cancelled
2 - Error
The RequestResponse enum however, has "kUnknown" as 0, and thus
"kSuccess" as 1 (with all other values also shifted up by 1 value). This
means that when the portal was cancelled, we were still receiving
RequestResponse::kSuccess. This fixes the issue by removing the improper
cast and adding a translation function. This function is local for now
since no where else attempted to cast values to a RequestResponse; but
can be moved if the need arises.
Fixed: chromium:1351824
Change-Id: I4cd44d90055147c9592d590c7969dcfc3297a3d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271240
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37755}
This combines the below IPv6 fixes into the field trial
WebRTC-IPv6NetworkResolutionFixes:
1. Prefer global IPv6 address over link local
2. Use address family when resolving STUN hostname
WebRTC-PreferGlobalIPv6ToLinkLocal is currently in Dev but will be
rolled back temporarily.
Bug: webrtc:14334, webrtc:14131
Change-Id: I1fb3f55c4c5f3c5c0b441ece30e72cf393e074d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271340
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37754}
The current behaviour is to lookup using AF_UNSPEC, which leaves
the decision up to the getaddrinfo implementation, then filter to
resolved addresses matching the network family anyway.
Looking up using the network's family upfront avoids resolving to
an unusable address.
Bug: webrtc:14319, webrtc:14131
Change-Id: I4997452dc26aeb82e5d2890701721e7d477803a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270625
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37753}
The input SocketAddress for STUN host lookup is constructed with just
the hostname, so the family is AF_UNSPEC. So added an overload with a
target family to distinguish this from the family of the input addr.
Bug: webrtc:14319, webrtc:14131
Change-Id: Ia5ac5aa2e894e0c4dfb4417e3e8a76a6cec3ea71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270624
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@google.com>
Cr-Commit-Position: refs/heads/main@{#37750}
that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.
Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
Resolves an issue where, in Chrome, WebRTC event logs do not capture outgoing packets for video receivers because no reference to the event log was passed to the video receiver.
Bug: webrtc:14338
Change-Id: Ia33ce6f2d69a0e341530648b10a08516dc53abf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271080
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37746}
Remove check if `prev_estimate_` is less than 10 us since it can never
be less than 1 ms.
Bug: None
Change-Id: If151390d22fa0b6ecdc36af64168d3e2049c7b6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271203
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37745}
Drops frames if the encoder has been configured with a new set of rtp
streams and a stray frame is returned from an encoder. This can happen with
hardware encoders that may deliver frames on a separate thread than were
they are configured.
This cl disable sending media on the RTP module a video layer is connected to and there by, old frames are dropped.
Bug: webrtc:1200, b/201798527
Change-Id: Id6bcfc3a846f6b8ed3b645cbbde571b819611a75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271122
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37744}
This behaviour has been fixed with the introduction of FrameBuffer3
Bug: webrtc:14033, webrtc:13343, webrtc:9974
Change-Id: Iba81c169706336e85194ed141324466e44a2c859
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265867
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37742}
SendStunBindingRequest can cause a SocketAddress pointed to by the
iterator to be deleted asynchronously before returning, causing `it`
to be invalid when incrementing in the continuation step.
Bug: webrtc:7309, webrtc:14131
Change-Id: I3f7d3d7c12935d9592ef3642679a821c58826df9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270744
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37741}
Also refactored a bit to support IPv6 networks and socket factories with
a mocked DNS resolver.
Bug: webrtc:14319, webrtc:14131
Change-Id: I32ac5beb9a72201bf83aac26aed6a670ed2d4955
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270741
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#37737}
Base port already keeps an AlwaysValidPointer to field_trials, so
TURN port's duplicate, private copy is redundant.
Bug: webrtc:14319, webrtc:14131
Change-Id: I94ee78ca5140c0b67826fbb94c35e28f30add943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270627
Reviewed-by: Jonas Oreland <jonaso@google.com>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37735}
Using ms() was converting the value to an int before putting it into a
double, causing the microseconds to be dropped. This has the most impact
on decode time metrics which are ofter less than 1ms.
Bug: webrtc:14339
Change-Id: Ie8401ba5a46eb3b35e8a699acfdad2dcd32a8240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271163
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37734}
Calling InitFieldTrialsFromString modifies a global variable so we must
make sure that state is reset between test runs.
Bug: webrtc:10335, webrtc:14336
Change-Id: Ia9839dd16a330ed3220ed470c28c541fc1cc0678
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271022
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37731}
If the source image has a native handle and the encoder supports
the native handle, the encoder is expected to be able to correctly
sample/scale the source.
And VTCompressionSession can handle this, so DCHECK the frame
resolution only if the frame buffer is not native.
Bug: webrtc:14318
Change-Id: Id19c2f3bd86e9a2e1034d20e0255b4adc04a781f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270144
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37730}
The data that's used to report the histograms is owned by UlpfecReceiver
and moving the reporting there, simplifies things as configuration
changes happen in RtpVideoStreamReceiver2 (which currently require all
receive streams to be deleted+reconstructed).
Additional updates:
* Consistently using `Clock` for timestamps. Before there was
a mix of Clock and rtc::TimeMillis.
* Update code to use Timestamp and TimeDelta.
Bug: none
Change-Id: I89ca28ec7067a49d6b357315ae733b04e7c5a2e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271027
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37729}
replace std::deque implementation with a manually controlled circular buffer.
replace Timestamp validity check from 'IsInfinite()' accesser to cheaper comparison to zero.
These greatly increase PacketArrivalTimeMap::AddPacket perfomance when packet arrive with large sequence number gaps.
Bug: chromium:1349880
Change-Id: I6f4e814b1086ca9d0b48608531e3a387d9e542dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270564
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37722}
The intention of this change is to separate the Kalman filter state
(that prior to this change lived in JitterEstimator) from the
other filter's state, making it easier to see how the different
filters interact.
This move does not include any interface, functional, or
documentation changes. Those will follow in later changes.
A very basic unit test is added, which will also be expanded
later on.
Bug: webrtc:14151
Change-Id: Ifb9b8ce2d9418ea52ccf64a77fd46d1ebba30779
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264984
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37721}
We pass the fd we recieve from xdg-desktop-portal to PipeWire to connect
to it and according to the specification PipeWire automatically closes
it on disconnect or failure. We also close the fd ourself when we tear
down the portal connection so we have to avoid doing this twice. Looks
OBS studio just duplicates the fd passed to PipeWire so do the same in
order to avoid the fd ownership violation once we stop sharing.
The fd we recieve from xdg-desktop-portal is from PipeWire also using
fcntl() with F_DUPFD_CLOEXEC option.
Bug: chromium:1339236
Change-Id: Ia7aee36e520dd5ff9a40688a6807e31c4e636f8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270421
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37712}
We only need to see which bitrates have been configured, no need to
wait for failed frame. This should also reduce test durations somewhat.
Bug: None
Change-Id: Ie081310f9f80e21039c78d8c80510769cb400c3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270747
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37711}
* Add test to Generic decoder unittests to ensure drop behaviour is covered.
* Use simulated time in the generic decoder unittests.
Bug: webrtc:14324
Change-Id: I10b28b45c434f92d5344683fb9ca6676efe0e08c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270662
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37710}
This replaces use of RunLoop and SimulatedClock. As a related change,
units like TimeDelta and Frequency are used as needed.
Bug: None
Change-Id: I892ee38641f2fd37d4bd1b0fb425bfb5d4706ac1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270626
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37708}
