The ObjC API (the files in sdk/objc/Framework/Headers/WebRTC/) needs to
be pure ObjC. The changes that are reverted here introduced C++ which
turns it into ObjC++.
We don't have a test protectcing this right now, but it's probably
something we should add to catch changes like this in the future.
TBR=alexnarest@webrtc.org,deadbeef@webrtc.org
Bug: webrtc:8243
Change-Id: Idea688f4014cd44c27cf2cb2a5ec8a9ea7da3c00
Reviewed-on: https://webrtc-review.googlesource.com/16429
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20463}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/15481.
This time with an extra (dummy) interface to ensure that we don't
break downstream clients.
Improves native Android audio implementations.
Bug: webrtc:8453
Change-Id: I659a3013ae523a2588e4c41ca44b7d0d2d65efb7
Reviewed-on: https://webrtc-review.googlesource.com/16425
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20462}
Always enabling verbose mode means about 100% more text is printed,
but this should not be a problem as the only time that we explicitly
look at the logs is when the bots are failing, or when we want to save
all output for plotting.
BUG=webrtc:8448
Change-Id: Ia5feab5220d047440d15cddb7d3fbca1c5a4aaf5
Reviewed-on: https://webrtc-review.googlesource.com/16140
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20461}
DEPS roll is blocked on it because it pins a broken clang version,
but Chromium has no commitment to fixing it.
Bug: chromium:774973
Change-Id: Id04fadde599293bca7b6c25faa2e9926c1265dc7
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/16423
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20460}
Always use the packetization formely known as kEqualSize.
The RTPFragementation header is ignored, which is no change
in behaviour, since the caller previously always passed null.
Bug: webrtc:6471
Change-Id: Id9e2f985280c2ee8cc33fcf0e5c1fc3ee61c1aff
Reviewed-on: https://webrtc-review.googlesource.com/15222
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20459}
In order to avoid conflicts with downstream projects WebRTC is going
to prefix its LOG macros with RTC_.
This CL renames all the LOG macros to macros with the RTC_ prefix and
it also defines backward compatibility LOG macros in order to let
downstream projects to switch to RTC_ prefixed macros without breaking
them.
A follow-up CL will remove the usage of LOG macros in WebRTC.
Bug: webrtc:8452
Change-Id: Ic3e495cba6c772f65259dc65ee278560a59d02d7
Reviewed-on: https://webrtc-review.googlesource.com/15442
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20456}
CGWindowID is 32-bit, WindowId is 64-bit, using WindowId to receive int value
from CFNumberGetValue() causes the top 32 bits to be random. WindowFinderMac is
impacted by this issue and returns a random number. WindowCapturerMac cannot
match the window_id_ with the the random number.
Meanwhile MouseCursorMonitorMac uses window title to match "Dock" window. See,
https://cs.chromium.org/chromium/src/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm?rcl=a194e58e799ccab6c999998e5d0f75725aa3f748&l=174
This logic should not be necessary on 10.12 or upper, the name of dock window
is not "Dock" anymore. But to ensure the consistency on old platforms, I have
also added this logic back into GetWindowList() function.
Bug: chromium:778049
Change-Id: Ie827bcd5d31f2ca69ff24c24cf640cb7cc50d419
Reviewed-on: https://webrtc-review.googlesource.com/15782
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20451}
This is done in preparation to make all javac warnings into errors for
WebRTC targets.
Bug: webrtc:6597
Change-Id: I402043157bd75943adf0de52111e5a1bb179c6d1
Reviewed-on: https://webrtc-review.googlesource.com/15104
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20450}
Apparently WebSocketObserver gets garbage collected if it is not stored
by us. This caused some external tests to break.
Bug: None
Change-Id: If62786e84f84a5a63172d67962bb4de8ae3e8479
Reviewed-on: https://webrtc-review.googlesource.com/16100
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20449}
Summary:
Adds AudioTrackStartErrorCode to separate different types of error
codes in combination with StartPlayout.
Harmonizes WebRtcAudioRecord and WebRtcAudioTrack implementations
to ensure that init/start/stop is performed identically.
Adds thread checking in WebRtcAudio track.
Bug: webrtc:8453
Change-Id: Ic913e888ff9493c9cc748a7b4dae43eb6b37fa85
Reviewed-on: https://webrtc-review.googlesource.com/15481
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20448}
This reverts commit 30915a742d86df55ac5c04501c0e8104675a612e.
Reason for revert: Breaks downstream.
Original change's description:
> Simple Default ObjC video codec factories.
>
> Move the simple video encoder/decoder factory from AppRTCMobile into the
> public API so users who don't have special requirements for video codecs
> can easily get started.
>
> Also clean up the API a little.
>
> This CL replaces the more flexible default factories in
> https://webrtc-review.googlesource.com/c/src/+/7741 and clients that
> want to implement their own codecs will have to supply their own
> encoder/decoder factories as well. The benefits of the approach in
> this CL are a simpler API and less effects on the rest of the code.
>
> Bug: None
> Change-Id: I4ed94090d778b4fc38b49864de1d4de4ff125d6a
> Reviewed-on: https://webrtc-review.googlesource.com/15141
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20441}
TBR=magjed@webrtc.org,andersc@webrtc.org,kthelgason@webrtc.org
Change-Id: I3d4395cc9667e6c6cdb33a3b0f5c5fb5bfde9028
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/15182
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20446}
I want to move away from the old encoder factory interface
cricket::WebRtcEncoderFactory to the new webrtc::VideoEncoderFactory. I
created a new webrtc::SdpVideoFormat that essentially is a subset of the
cricket::VideoCodec variables. E.g. the encoder factories shouldn't have
to assign payload types to the codecs, so the payload is not part of
webrtc::SdpVideoFormat. I also didn't add the "feedback_params" that is
used in cricket::VideoCodec to webrtc::SdpVideoFormat. This is causing
problems now, because the internal encoder factory is adding flexfec
feedback params. To avoid this problem, I add these feedback params in
WebRtcVideoEngine instead, like we do for the other codecs.
Bug: webrtc:7925
Change-Id: I7c6ae8d1e1f47f3631c4804c223ec21da8d73685
Reviewed-on: https://webrtc-review.googlesource.com/15223
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20444}
Adds a constructor for DirectTransport that takes a pointer to an instance
of a class derived from FakeNetworkPipe. Said class can override Process()
and SendPacket(...) members thereby emulating any desired network behavior.
Bug: b/67487983
Change-Id: I829fd3506124db61587af19192a14fdf62b06ca5
Reviewed-on: https://webrtc-review.googlesource.com/14620
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20443}
Move the simple video encoder/decoder factory from AppRTCMobile into the
public API so users who don't have special requirements for video codecs
can easily get started.
Also clean up the API a little.
This CL replaces the more flexible default factories in
https://webrtc-review.googlesource.com/c/src/+/7741 and clients that
want to implement their own codecs will have to supply their own
encoder/decoder factories as well. The benefits of the approach in
this CL are a simpler API and less effects on the rest of the code.
Bug: None
Change-Id: I4ed94090d778b4fc38b49864de1d4de4ff125d6a
Reviewed-on: https://webrtc-review.googlesource.com/15141
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20441}
Old CreateVideoDecoder interface is deprecated. This allows
VideoDecoderFactoryWrapper to create codecs for types that WebRTC
doesn't know about.
Bug: webrtc:8140
Change-Id: I69aa1a0164642b4e4377daa1abeb9039c04fd884
Reviewed-on: https://webrtc-review.googlesource.com/15401
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20438}
This CL lowers the threshold for delay change detection in AEC3.
This makes the delay decisions more stable.
TBR=gustaf@webrtc.org
Bug: chromium:778396,webrtc:8451
Change-Id: I8b015455399d696172b7c0beb033caf508f426e9
Reviewed-on: https://webrtc-review.googlesource.com/15541
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20433}
{root_x, root_y} should be used to report the absolute cursor position in
MouseCursorMonitorX11.
Bug: chromium:778035
Change-Id: I421005d52786a57da8e8c3901bdf4afa2843ff24
Reviewed-on: https://webrtc-review.googlesource.com/15680
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20432}
Some death tests for AEC3 cause memory leaks on trybots. This CL
temporarily disables BlockProcessor.VerifyRenderBlockSizeCheck.
Bug: webrtc:8449,webrtc:6985
Change-Id: I2900a73f7c7d5bf0e8b58a20f9a40bd5d654629a
Reviewed-on: https://webrtc-review.googlesource.com/15500
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20431}
This CL replaces 5 left shifts where the shifted value may be
negative. The shifts are replaced with equivalent multiplications.
Bug: chromium:777231, chromium:776719, chromium:776624, chromium:776286
Change-Id: Ifb27d5506eac779e60f238432bdf9e4bc5b2da4c
Reviewed-on: https://webrtc-review.googlesource.com/14800
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20430}
When audio_processing_fuzzer runs with 'DCHECK_ALWAYS_ON', it crashes
when both AEC and AECM is enabled at the same time. This change
detects that case and fixes
https://clusterfuzz.com/v2/testcase-detail/6389429496446976.
It also removes an unnecessary safeguard that didn't allow fuzzing
with 8kHz input signals.
Bug: chromium:776358
Change-Id: I33c18a2a235e50ae410f7be24637872823e432eb
Reviewed-on: https://webrtc-review.googlesource.com/15320
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20428}
Unlike all the other API directories, api/ is the root of an entire
tree of directories that are also API directories.
BUG=webrtc:8445
Change-Id: I218befe6fb6113b95599512f062ebe63abc98889
Reviewed-on: https://webrtc-review.googlesource.com/15321
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20427}
We've done this previously with the other audio encoders, but Opus had
to wait until all external users had been updated.
BUG=webrtc:7847
Change-Id: I70422d7b6c715f32a43bee88febcf6b6155e18b3
Reviewed-on: https://webrtc-review.googlesource.com/8000
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20424}
Add fine grained dropped video frames counters on sending side
4 new counters added to SendStatisticsProxy and reported to UMA and logs.
Bug: webrtc:8355
Change-Id: I1f9bdfea9cbf17cf38b3cb2f55d406ffdb06614f
Reviewed-on: https://webrtc-review.googlesource.com/14580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20421}
Now decision between using SimulcastEncoderAdapter and using VP8 encoder
is postponed before codec is initialized for VP8 internal codecs. This is done
be new VP8EncoderProxy class. New error code for codec initialization is used
to signal that simulcast parameters are not supported.
Bug: webrtc:7925
Change-Id: I3a82c21bf5dfaaa7fa25350986830523f02c39d8
Reviewed-on: https://webrtc-review.googlesource.com/13980
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20419}
This CL balances the NLP tradeoff in AEC3 to properly handle the cases
when the echo path is so strong that it saturates the echo and when it
is so weak that the echo is very low compared to nearend.
Bug: webrtc:8411, webrtc:8412, chromium:775653
Change-Id: I5aff74dfadd51cac1ce71b1cb935d68a5be6918d
Reviewed-on: https://webrtc-review.googlesource.com/14120
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20418}
system_wrapper/Atomic32 has been deprecated (which is already just a
wrapper of std::atomic) in favor of platform-independent std::atomic
from C++11. This CL replaces all use of Atomic32 in video/
Bug: webrtc:8428
Change-Id: If4dab4909df06944c009e7b70141f58daef7be10
Reviewed-on: https://webrtc-review.googlesource.com/14720
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Yuwei Huang <yuweih@google.com>
Cr-Commit-Position: refs/heads/master@{#20417}
