Isac is not able to effectively compress all types of signals. This
should be extremely rare with real audio signals, but mostly happens
with artificially created test signals. When this happens, we should
avoid crashing and just carry on.
Bug: chromium:1170167
Change-Id: I97b551fbbdcccb0186f3e6497991ac52d2301f68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205626
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33193}
When calculating maximum allowed size for a media packet.
In particular take in account that rtx packet might need to
include mid and repaired-rsid extensions when media packet can omit them.
Bug: webrtc:11031
Change-Id: I3e7bc36437c23e0330316588d2a46978407c8c45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206060
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33184}
This CL adds the ability in audioproc_f and unpack_aecdump to:
-Clearly identify the Init events and when those occur.
-Optionally only process a specific Init section of an aecdump.
-Optionally selectively turn on dumping of internal data for a
specific init section, and a specific time interval.
-Optionally let unpack_aecdump produce file names based on inits.
Bug: webrtc:5298
Change-Id: Id654b7175407a23ef634fca832994d87d1073239
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196160
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33181}
Since for such frame SvcController haven't setup how buffer should be
referenced and updated, the frame would likely have unexpected configuration.
Log an error to note resource have been wasted produce it and drop such frame.
Bug: webrtc:11999
Change-Id: I1784403e67b7207092d46016510460738994404e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205140
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33148}
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.
`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.
Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
To be able to build these targets in chromium we need to replace all abseil dependencies with "//third_party/abseil-cpp:absl".
Bug: webrtc:12404
Change-Id: Ie0f6af73f2abc73e5744520cfd9a6414e2f948e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202762
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33108}
This is a follow up to
https://webrtc-review.googlesource.com/c/src/+/200161
I forgot to actually upload the code that addressed reviewer's comments,
and since they were minor comments I went ahead and completed the CL
before I realized.
Fortunately no harm done because all of this code is behind a disabled
build flag. This is a great example of move fast, make mistakes. Lesson
learned.
Bug: webrtc:9273
Change-Id: I5e35b31719b264855568de4b5e595fe0f192654e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204420
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33095}
Some locations in the WebRTC codebase RTC_LOG the value of the
__FUNCTION__ macro which probably is useful in debug mode. Moving
these instances to RTC_DLOG saves ~10 KiB on arm64.
Bug: webrtc:11986
Change-Id: I5d81cc459d2850657a712b9aed80c187edf49a3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203981
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33086}
This change refactors WgcWindowCapturer into WgcCapturerWin, a source
agnostic capturer, and finishes the implementation to enable both window
and screen capture.
This CL depends on another which must complete first:
196622: Add ability to load CreateDirect3DDeviceFromDXGIDevice from
d3d11.dll | https://webrtc-review.googlesource.com/c/src/+/196622
This feature remains off by default behind a build flag, due to it
adding a depency on the Win10 SDK vv10.0.19041 which not all consumers
of WebRTC have upgraded to. A follow up change later will enable the
rtc_enable_win_wgc build flag, but for now it should remain off.
The basic operation of this class is as follows:
Consumers call either WgcCapturerWin::CreateRawWindowCapturer or
CreateRawScreenCapturer to receive a correctly initialized
WgcCapturerWin object suitable for the desired source type.
Callers then indicate via SelectSource and a SourceId the desired
capture target, and the capturer creates an appropriate WgcCaptureSource
for the correct type (window or screen) using the
WgcCaptureSourceFactory supplied at construction.
Next, callers request frames for the currently selected source and the
capturer then creates a WgcCaptureSession and stores it in a map for
more efficient capture of multiple sources.
The WgcCaptureSession is supplied with a GraphicsCaptureItem created by
the WgcCaptureSource. It uses this item to create a
Direct3D11CaptureFramePool and create and start a
GraphicsCaptureSession.
Once started, captured frames will begin to be deposited into the
FramePool. Typically, one would listen for the FrameArrived event and
process the frame then, but due to the synchronous nature of the
DesktopCapturer interface, and to avoid a more complicated multi-
threaded architecture we ignore the FrameArrived event. Instead, we
wait for a request for a frame from the caller, then we check the
FramePool for a frame, and process it on demand.
Processing a frame involves moving the image data from an
ID3D11Texture2D stored in the GPU into a texture that is accessible
from the CPU, and then copying the data into the new WgcDesktopFrame
class. This copy is necessary as otherwise we would need to manage the
lifetimes of the CaptureFrame and ID3D11Texture2D objects, lest the
buffer be invalidated.
Once we've copied the data and returned it to the caller, we can unmap
the texture and exit the scope of the GetFrame method, which will
destruct the CaptureFrame object. At this point, the CaptureSession
will begin capturing a new frame, and will soon deposit it into the
FramePool and we can repeat.
Bug: webrtc:9273
Change-Id: I02263c4fd587df652b04d5267fad8965330d0f5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200161
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33083}
As an alternative to attaching custom array of bytes.
Bug: b/178094662
Change-Id: I92dcbf04998d8206091125febc520ebfcc4bcebf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203264
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33069}
Reduce amount of dynamic memory used to generate rtcp message
Remove option to request several types of rtcp message as unused
Deduplicated PacketContainer and PacketSender as constructs to create packets
Bug: None
Change-Id: Ib2e20a72a9bd73a441ae6b72a13e18ab5885f5c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203261
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33068}
The SpatialIndex value from an EncodedImage is 0-based, but values were
off by 1.
Bug: none
Change-Id: Ie74e6450ddef1cfaee68fa230c441030fa86a64a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203525
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33067}
To make VideoCodec::scalability_mode the only option to set and
change the scalability structure, for easier maintainability.
Bug: webrtc:11404
Change-Id: I6570e9a93ddf2897ff7584c5d20a246346e853e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192361
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33056}
Finding the array element with the largest argmax is a fairly common
operation, so it makes sense to have a Neon optimized version. The
implementation is done by first finding both the min and max value, and
then returning whichever has the largest argmax.
Bug: chromium:12355
Change-Id: I088bd4f7d469b2424a7265de10fffb42764567a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201622
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33052}
Using WebRTC-VP9-PerformanceFlags and settings a multi-layer config,
and then configuring the codec in non-svc mode would cause us to not
set the cpu speed in libvpx. For some reason, that could trigger a
crash in the encoder.
This CL fixes that, and adds new test coverage for the code affected
byt the trial.
Bug: chromium:1167353, webrtc:11551
Change-Id: Iddb92fe03fc12bac37717908a8b5df4f3d411bf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202761
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33051}
This CL allows separate dump sets to be used when dumping internal
APM data using audioproc_f, opening up for reducing the amount of
data to be dumped.
Bug: webrtc:5298
Change-Id: I8286933ceed10db074f2064414cc08e8b12653fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196089
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33047}
To determine the appropriate amount of shifting to prevent overflow in a
cross correlation, it is necessary to have the max value of both
sequences. However, only one was calculated in the ilbc code. This CL
calculates the max of the other sequence and correctly takes both into
account.
Bug: chromium:1161837
Change-Id: I3ba8eee0814bb5eda3769c0ce6caf2681c7525e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202253
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33043}
Also making some more state const.
Instances of this class are currently constructed and used on the
"worker thread" but as part of the work for bug webrtc:11993, the
instances will be moved over to the network thread. Since the
class as is does not require synchronization, that is a good property
to make explicit now and then make sure we maintain it in the
transition.
Bug: webrtc:11993
Change-Id: Id587a746ce0a4363b9e9871ae1749549f8ef3e24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202681
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33035}
- The injection of the AGC2 level estimator into `AgcManagerDirect`
is not used anymore
- `ExperimentalAgc::enabled_agc2_level_estimator` can also be removed
- 3 ctors of `ExperimentalAgc` are unused
- `AgcManagerDirectStandaloneTest::AgcMinMicLevelExperiment` can be
split into separate unit tests (better code clarity)
Bug: webrtc:7494
Change-Id: I5843147c38cf7cb5ee484b0a72fe13dcf363efaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202025
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33027}
Moves the deprecated version of RtpRtcp module, and related classes
in video/.
Bug: webrtc:11581
Change-Id: Icc4cedb844fcd7c7372e8a907e5252f5b4fd955e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196904
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33025}
by updating flag that T1 frame can be referenced when it is encoded
rather than when it is sent for encoding.
Otherwise when encoder drops T1 frame, configuration for following T2 frame would
still try to reference that absent T1 frame leading to invalid references.
Bug: None
Change-Id: I6398275971596b0618bcf9c926f0282f74120976
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202030
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33002}
This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a
Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
