If at creation of a VP8 encoder there is not enough bitrate to enable a
given spatial layer - the configuration won't be updated to indicate
the correct temporal layer count. This means GetEncoderInfo() will
indicate lack of temporal layer support, which triggers issues with
rate allocation.
This CL fixes that by always setting an initial bitrate of 0bps.
Bug: webrtc:12788
Change-Id: I10974e85446b58e597d2ca415eaf2550306ce986
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220929
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34198}
Enumerating windows owned by the current process on Windows has some
complications due to the GetWindowText*() APIs potentially causing a
deadlock. The APIs will send messages to the window's message loop, and
if the message loop is waiting on this operation we will enter a
deadlock.
I previously put in a mitigation for this [1] which brought the
incidence rate down by an order of magnitude, but we are still seeing
this issue fairly frequently.
So, I've added DesktopCaptureOption enumerate_current_process_windows
which allows consumers to avoid this issue completely by ignoring
these potentially problematic windows.
By default the flag is set to true which equates with the current
behavior, consumers can set the flag to false to get the new behavior.
I've also updated all the capturers that enumerate windows on Windows
to respect the option.
[1] https://webrtc-review.googlesource.com/c/src/+/195365
Bug: chromium:1152841
Change-Id: I0e0d868957d6fbe1e607a440b3a909d005c93ccf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219380
Commit-Queue: Austin Orion <auorion@microsoft.com>
Reviewed-by: Joe Downing <joedow@chromium.org>
Cr-Commit-Position: refs/heads/master@{#34191}
We need to specify that the cursor should be included in the stream as
by default xdg-desktop-portal defaults to hidden cursor.
Bug: chromium:1202526
Change-Id: Ic4742da2e51f7ed28cb9d7b6b0c069c1fa7d0cee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214782
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34137}
Since all test cases that used RtpSenderEgress have been refactored or
moved, we can now get rid of lot of test fixture crud:
* Remove RtpSenderContext helper, make sender normal member.
* Remove test transport helper
* Remove task queue helper (needed for thread checks in egress)
* Remove various mocks no longer used
* Remove RtpSenderWithoutPacer subclass
* Remove WithWithoutOverhead parametrization (only affect egress)
..plus some cleanup of how configs are created.
Bug: webrtc:11340
Change-Id: I5c581d60862fc6dc2b99f76058782309dc7aef4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220280
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34135}
Thus removing dependency on RtpSenderEgress, allowing simplification of
the test fixture in a follow-up.
Bug: webrtc:11340
Change-Id: I9772bab18d1f4a04e0deccc9125d4b1c16c30d7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219627
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34132}
Reland of commit a743303211b89bbcf4cea438ee797bbbc7b59e80
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.
In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.
Further changes:
- If RTP header encryption enabled, prefer encrypted extensions over
non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
is not supported for that extension
- Mark FindHeaderExtensionByUri without filter argument as deprecated
Bug: webrtc:11713
Change-Id: I52a5ade1b94bc01d1c2a35cb56023684fcaf9982
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34129}
This CL makes a number of test use the paced sender callback to verify
the output of RTPSender, instead of re-parsed data from RtpSenderEgres.
Bug: webrtc:11340
Change-Id: I13ccf5a5db4b6df128cf2fa9e8dad443fcd15cdd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220162
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34126}
Set we don't use full stream width. This follows same code as in PW 0.3
case, it was just accidentally omitted.
Bug: chromium:682122
Change-Id: Ifb9200a14387ba9b9da3246c9c4e30306393c4e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214700
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Wez <wez@google.com>
Cr-Commit-Position: refs/heads/master@{#34124}
These should be the last of the testis from rtp_sender_unittest.cc that
should be moved and refactored to just test RtpSenderEgress.
Bug: webrtc:11340
Change-Id: Id09d7bbade608dd7194dcd8843d4f2887842a372
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220140
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34118}
These tests were likely made back when PacketRouter was iterating over
the RTP modules to find the correct to send on. Now that this is just
a DCHECK, it's already implicitly covered by other tests that actually
test the respective packet type functionality. Let's thus just remove
these old tests.
Bug: webrtc:11340
Change-Id: I244ca7e365378f4e48a601464b5df0e1d07732be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219621
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34116}
This CL refactors RtpSenderTest.SendPacketHandlesRetransmissionHistory,
moves some testing to rtp_ender_egress_unittest and adds test coverage
for a few cases.
Bug: webrtc:11340
Change-Id: Ic225d2af43c3926f69fe3ea45f41b18c29b8b4fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219796
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34111}
This check was important when header bytes were copied from source
packet to destination, but current implementation (new line 123) slices
the source packet, making capacity of the destination packet irrelevant.
Bug: b/189015462
Change-Id: I7e649cb7dfc6ba0fbe989c943e6515ab0da05fef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219695
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34110}
Instead, cap the final bandwidth estimate by the last received cap. This allows fast rampup after a REMB cap is lifted.
Bug: webrtc:12306
Change-Id: Ia99707134ce145275460524b3e46923876fdf62f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219696
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34109}
ffmpeg is going to be hiding the implementation of AVPacket, so we can't
allocate them on the stack anymore. av_init_packet is marked deprecated
on TOT ffmpeg, so remove its use everywhere in favor of av_packet_alloc
and av_packet_free.
Bug: chromium:1211508
Change-Id: I154311071123110dd749c71dec1ec2a0452b3908
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217780
Commit-Queue: Ted Meyer <tmathmeyer@google.com>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34106}
This is not defined anywhere, including chromium.
Bug: None
Change-Id: If5e89880570a80dd5720e48ebaefb0eb2c37fab3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215360
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#34105}
This is loosely based on the similar implementation in gecko.
Bug: webrtc:9965
Change-Id: I5203a05e1c34ca6f97bd1b143790f95ff245e340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219791
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Doudou Kisabaka <doudouk@google.com>
Cr-Commit-Position: refs/heads/master@{#34102}
This simplifies some tests and removes dependency on RtpSenderEgress
for those tests in rtp_sender_unittest.
Bug: webrtc:11340
Change-Id: I37489875947b0ac48a1742d2e9945510ee002f99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219624
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34099}
This consolidates the querying of the Ntp time in once place, the clock.
Bug: webrtc:11327
Change-Id: I14b19c2380996571d8c67c2c186629c209787162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219794
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Paul Hallak <phallak@google.com>
Cr-Commit-Position: refs/heads/master@{#34083}
to NTP.
No-Try because of lack of infra lack of capacity on macs.
No-Try: True
Bug: webrtc:11327
Change-Id: Ie0c9983031a6d37ae54b1d2381c229bee1a89e8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214134
Commit-Queue: Paul Hallak <phallak@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34078}
This allows callers to use timestamps generated from their own clocks
without worrying about converting to webrtc time.
No-Try because of lack of infra lack of capacity on macs.
No-Try: True
Bug: webrtc:11327
Change-Id: I7b1935654a2b23cf844c7b3622ed68763ced9da5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219785
Commit-Queue: Paul Hallak <phallak@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34076}
This also changes the meaning of |estimated_capture_clock_offset| in
|absolute_capture_time_| to become a remote to capturer clock offset.
Bug: chromium:1056230, webrtc:10739
Change-Id: Id658590e027bbe77ae0834ea224e1dc977a305f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219163
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#34067}
This change enables the use of two different adaptation speeds of the
matched filter of the delay estimator of AEC3.
One speed is used when no delay has been found, and one is used after a
reliable delay has been found. The purpose is to use a slower adaptation
speed to reduce the risk of divergence during double-talk without
slowing down the search for the initial delay.
The CL prepares for experimentation by adding field trials for
controlling the two adaptation speeds.
Bug: webrtc:12775
Change-Id: I817a1ab5ded0f78d20de45edcf04c708290173fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219083
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34055}
In CL https://webrtc-review.googlesource.com/c/src/+/216323 we fixed
the issue where I420 and I420A not being equal would result in dropping
frames in release builds.
But we forgot to update the corresponding DCHECK, meaning the I420 not
being the same as I420A issue still causes crashes on debug builds.
(I must have been running a release build not to catch this before?)
This CL replaces the DCHECK_EQ with an RTC_NOTREACHED inside the
IsCompatibleVideoFrameBufferType check.
Because this only affects debug builds, this CL does not need to be
backmerged anywhere.
Bug: chromium:1203206
Change-Id: I101823e8bca293e94d0f7ce507fe78cedff3ea1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219281
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34048}
This is a follow-up to r34019. It adds checks for when padding can be
sent before media - and how timestamps are set on RTX padding.
Bug: webrtc:11340
Change-Id: I46fbd3c3eff9e308b5c65220718df749f2d9c46b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219162
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34041}
This change adds the field trial "WebRTC-TransientSuppressorForcedOff"
that can be used to disable the transient suppressor (removal of
keyboard typing sounds). The field trial can be enabled by users via
command-line or via experimentation.
Bug: chromium:1186705
Change-Id: I7272df6a20fbbee24a7ba0904502c76bd775d275
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219282
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34038}
