This brings the two ProcessStream functions closer in implementation.
Additionally, the error checking that is currently done in the period of not holding the lock seems cheaper than releasing and reacquiring the capture lock.
Bug: webrtc:11235
Change-Id: Ib4afc68afb419fcabbb8cf08a3a2e61d2c12acda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163021
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30140}
This CL ensures that the AGC2 is created and initialized only when
needed.
Apart from that, the CL also avoids a runtime-reallocation that happens
each time the setting is applied.
Bug: webrtc:5298
Change-Id: Iad9eaa05a3d0baa0788cd11b2aa17ddd8e0c509b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163987
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30139}
This CL moves the activation of the transient suppression to the APM
config.
Bug: webrtc:5298
Change-Id: Iba7975bec4654c3df8834fd5b7d1082ff53641dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163985
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30137}
Replace all usages of java_files with sources in gn files, and
automatically format.
This is in preparation for java_files being completely removed upstream
in favor of sources.
NOPRESUBMIT=true
Bug: chromium:1035074
Change-Id: Ib9a698740b7ad26a127031d90321c7ae2feb59bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Natalie Chouinard <chouinard@google.com>
Cr-Commit-Position: refs/heads/master@{#30135}
This CL changes the GetStatistics call in the audio processing module
(APM) to not aquire the render or capture locks in APM, while still
being thread-safe.
This change eliminates the risk of thread-priority inversion due to the
GetStatistics call.
Apart from the above the CL:
-Corrects the GetStatistics to not be const (it was const even though it
aquired locks).
-Slightly changes the statistics reporting, so that the stats received
may be older than the most recent stats reported.
Bug: webrtc:11241
Change-Id: I00deb5507e004cbe6e4a19a8bad357491f86f4ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163982
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30131}
This CL allows the noise suppressor to use the linear AEC output
for analysis whenever that is available. This will potentially
lower the risk that the noise suppressor estimates the noise
based on echo.
The feature is off by default.
Bug: webrtc:5298,b:132164318
Change-Id: Idc6c8e197d96209d213819d87a8fb2533b7303ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162900
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30116}
This CL ensures that the no data vectors in the reverb computer code
are fixed. This allows arbitrary long filters to be used, and ensures
that a minimum required heap size is used.
Bug: webrtc:8671
Change-Id: I7085ed262a3f5965d796270434b6578f4030606e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162661
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30115}
This CL moves the high-pass filter to run in the full-band domain
instead of the split-band domain.
Bug: webrtc:11193
Change-Id: Ie61f4a80afda11236ecbb1ad544bbd0350c7bbfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161453
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30112}
The pacing controller allowed sending bitrate probes, despite it being
paused. This CL adresses that, and makes sure the task-queue based mode
also properly repsects pausing.
Bug: webrtc:10809
Change-Id: I79643c9a24666110d7583fce3ed1bfd6865e9e10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162520
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30109}
If for some reason capture timestamp is unset, the default value of 0 would be
passed to RtcpSender. This will cause rtp timestamps to grow at double the rate
in Sender Reports because it has time since the last frame capture as a term.
Bug: none
Change-Id: I2fe09dabef6b0957fb504deaa06393dedc4a9e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162481
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30105}
It's not great to use asserts in util functions like this because it
breaks the arrange-act-assert rule, but using checks is worse because
they will crash the test process on failure (= no other tests get run
after that).
Bug: b/143587130
Change-Id: If4d085311de0792b9fca1584db299fd24199e72e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162360
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30093}
to return time of the last receieved packet of a key frame rather than
last received first packet of a key frame.
To match VideoReceiveStream expectation and prevent requesting
a new key frame if a large key frame is currently on the way.
Bug: None
Change-Id: I443a60872a3580d324f050080a9868f7b90d71a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161730
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30084}
USE_X11 is magically set by the toolchain. Let's tie this to
whether X11 extensions is on or not. WEBRTC_USE_X11 is used
in the audio device module so let's be consistent with that.
Bug: b/143587130
Change-Id: I15b1744c9acb534dbdf2119deb2dc5d17aa04184
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161901
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30080}
from LS_INFO to LS_VERBOSE.
By default, unit tests run with logging at info level.
A random run today produced more than 70.000 lines of
output. This CL would reduce that by approximately 15.000.
Bug: none
Change-Id: Ie62708cebf109510a2443aa5ab5c4e645ffc6707
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161950
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30077}
This prepares for moving TransportFeedbackAdapter to TaskQueue.
Bug: webrtc:9883
Change-Id: Ib333f6a6837ff6dd8b10813e8953e4d8cd5d8633
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162040
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30076}
Some clients will not count audio packets into the bandwidth estimate
despite negotiating e.g. abs-send-time for that SSRC.
If padding is sent on such an RTP module, we might get stuck in a low
resolution.
This CL works around that by preferring to send padding on video SSRCs.
Bug: webrtc:11196
Change-Id: I1ff503a31a85bc32315006a4f15f8b08e5d4e883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161941
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30066}
This reverts commit 60ec3703cd1f87081c0e4becde5d9ef210a6d44a.
Reason for revert: Needs back-end test before always enabling.
Original change's description:
> Ensure loss-based controller is always enabled.
>
> The new default parameters are the ones that were used in the Chrome
> Finch trial. The deleted unit test is invalidated by these changes.
>
> Bug: chromium:941413
> Change-Id: I597f4b0defaebe5bb3a6710b071fae2ee5c6f461
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160652
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30049}
TBR=srte@webrtc.org,crodbro@webrtc.org,jonasolsson@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:941413
Change-Id: I5da4676ad8be2569ad7eed99e954e0d0b624110b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161902
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30061}
This CL removes the remaining settings for using the legacy AEC.
It also adds a missing printout of the enforce_high_pass_filtering
parameter in the ToString method.
Bug: webrtc:11165
Change-Id: I58f0861bf1c6cd24bd83f4d3e394653b2fab3d71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161683
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30050}
The new default parameters are the ones that were used in the Chrome
Finch trial. The deleted unit test is invalidated by these changes.
Bug: chromium:941413
Change-Id: I597f4b0defaebe5bb3a6710b071fae2ee5c6f461
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160652
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30049}
As of https://webrtc-review.googlesource.com/c/src/+/158899, FEC may be
used on packets with VideoTimingExtension. This may result in creation
of FEC packets that exceed the maximum configured RTP packet size.
This problem occurs most frequently with datagram transports that define a
smaller maximum packet size.
Bug: webrtc:9719
Change-Id: I842216a6696a695f0a3f01a221e538605fc5b9bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161557
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30045}
This CL removes the deprecated legacy AEC code.
Note that this CL should not be landed before the M80 release has been cut.
Bug: webrtc:11165
Change-Id: I59ee94526e62f702bb9fa9fa2d38c4e48f44753c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161238
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30036}
This is a reland of a0adf3d4409036d095480e9bfa0fc06990362f84
Original change's description:
> Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."
>
> This is a reland of e7153012682ccd3d1eacc18f802cab7820e3bad3
>
> Original change's description:
> > Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
> >
> > Bug: chromium:396091
> > Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768
> > Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
> > Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
> > Cr-Commit-Position: refs/heads/master@{#29083}
>
> Bug: chromium:396091
> Change-Id: I0d9171ae5f340e0489e4b45ce5d97bc52b0a4904
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156067
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29655}
Bug: chromium:396091
Change-Id: I47525911095fabc6cee613d03b0d83134b95b084
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158900
Reviewed-by: Tomas Gunnarsson <tommi@chromium.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30032}
while suboptimal, these implementions are complete and allow to
swap code from using RtpDepacketizer interface to VideoRtpDepacketizer
Bug: webrtc:11152
Change-Id: Ie7823feeb5b0563b74754255aaedfada9d446ac5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161380
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30031}
This removes some code in the AudioDeviceWindowsCore::CoreAudioIsSupported function that was checking that every audio input and output device was functional. There are legitimate cases where some, or all, audio devices may not be accessible, and that was causing CoreAudioIsSupported to return false.
If CoreAudioIsSupported returns false, a subsequent RTC_CHECK call fails, which causes the entire app to exit.
After this change, the CoreAudioIsSupported() function simply checks if the Core Audio APIs are supported and no longer tries to do extra stuff unrelated to checking if the APIs are supported.
Note that Core Audio is actually supported in all versions of Windows after Windows XP. There were log messages in the code saying that if CoreAudioIsSupported() returns false, WebRTC will use the Wave Audio APIs instead. But this is no longer the case. The Wave Audio APIs would only be needed for Windows XP, and this code appears to have already been removed from WebRTC.
It is tempting to simply make CoreAudioIsSupported() do a "return true;" but for now I only removed the part of the logging messages that mentioned the Wave Audio APIs.
I understand that there is a new Audio Device Module (ADM) called WindowsCoreAudio2, which is now recommended for use by apps. Apps are supposed to instantiate WindowsCoreAudio2 and pass it in to WebRTC. When the app supplies its own ADM, CoreAudioIsSupported() does not get invoked, which avoids the bug. To help make it clearer that using WindowsCoreAudio2 is an acceptable solution, I am removing a comment that says that kWindowsCoreAudio2 is "experimental".
Bug: webrtc:11081
Change-Id: I7ed1684a276799f4c83006b45629e48814f0b18b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161463
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30025}
This CL follows the "Rule of zero".
Those constructors made no sense compared to default generated ones,
since all members are POD.
They were introduced to quiet a memory sanitizer warning,
which apparently was misleading.
As a bonus, the struct is now movable.
Bug: webrtc:11180, webrtc:9855
Change-Id: Iff9fd950bec8040bc6e7e7ece33cc49c5f453f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161381
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30023}
This change avoids inadvertent capture of certain system windows (e.g.
the Start menu, other taskbar menus, and notification toasts) when
capturing a specific window on Windows.
It stops using EnumWindows for detection of overlapping windows, because
this API excludes these system windows from its enumeration. Using
FindWindowEx instead enumerates these windows.
The enumeration logic is refactored somewhat because a callback is no
longer necessary.
Bug: webrtc:10835
Change-Id: I1cccd44d6ef07f13a68e8daf2d2573d422001201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161153
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30022}
