Merges the two targets in modules/audio_processing
and removes some redundant code. This enables not writing
a bunch of redundant code in
https://webrtc-review.googlesource.com/c/src/+/70502
':audio_processing' did depend on ':aec_dump_interface'.
'modules/audio_processing/aec_dump' did depend on
'aec_dump_interface' but not ':audio_processing'.
Having the AecDump implementation not depending on
'audio_processing' allows to have faster compilation time and
reduces the dependencies. However, maintaining such a decoupling
makes APM and AecDump client code more complex.
NOTRY=true # want this in and 'ios_api_framework' seems stuck.
Bug: webrtc:7404
Change-Id: I75a5f234591014ac42d52bc1a36526072f5be89c
Reviewed-on: https://webrtc-review.googlesource.com/76603
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23244}
The current implementation carefully shifts down the energy so as not to overflow.
The fuzzer audio_decoder_ilbc_fuzzer found an integer overflow anyway.
The energy is only used for a threshold check.
This fix stops the energy computation when the threshold is reached, before it can overflow.
Bug: chromium:837922
Change-Id: I45e84d2d271a37e6476b08433a2cbd5a8f6e6f26
Reviewed-on: https://webrtc-review.googlesource.com/76122
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23242}
This CL simply renames the test cases that were not renamed in
prior CLs.
Bug: None
Change-Id: If616eb823e1453bc92ba037722b77a219d54409c
Reviewed-on: https://webrtc-review.googlesource.com/76780
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23240}
CustomProcessing is the interface to injectable audio processing
submodules to AudioProcessing. This CL makes it possible to set
runtime settings on the injected render processing component.
Note that the current runtime setting handling happens on the capture
thread. Therefore, we add another SwapQueue to communicate with the
render thread.
Bug: webrtc:9138, webrtc:9262
Change-Id: I665ce2d83a2b35ca8b25cca813d2cef7bd0ba911
Reviewed-on: https://webrtc-review.googlesource.com/76123
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23236}
This CL finalizes the feature extraction part for the RNN VAD adding
a class that combines a high-pass filter, LP residual computation,
pitch estimation and spectral features extraction.
This CL also includes a minor refactoring of the pitch estimation
library.
Bug: webrtc:9076
Change-Id: I918b9e143bc6dd2bf508a891446067258a68a777
Reviewed-on: https://webrtc-review.googlesource.com/75504
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23235}
This prepares for making the BBR implementation more identical to the
implementation in Quic, to ensure that results are comparable.
Bug: webrtc:8415
Change-Id: Icc66b748e52efb80b5c3ade85f83de6b224a4bd2
Reviewed-on: https://webrtc-review.googlesource.com/76581
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23227}
- Two quality layers (same resolution, different bitrate).
- Max bitrate of low layer is limited to 200kbps. The choice of the
limit is driven by VP8 screen sharing which limits max bitrate of low
temporal layer to 200kbps. Using the same value for VP9 guarantees
that there will be no regressions for participants with limited
bandwidth.
- Max bitrate of high layer is limited to 500kbps. According to test
results this value is enough to get up to +5dB higher PSNR than VP8
SS provides on 1.2Mbps (max total bitrate for VP8 SS) link.
- Max total sent bitrate is limited to 700kbps. It is 500kbps lower
than that in VP8 SS (1200kbps).
Bug: webrtc:9261
Change-Id: I7919cc3933064664567c39e380a44cad0c65f1e8
Reviewed-on: https://webrtc-review.googlesource.com/76380
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23226}
Now using attribute to ensure that we avoid error like these when bulding with -Wthread-safety-analysis:
error: mutex '_critSect' is still held at the end of function [-Werror,-Wthread-safety-analysis]
RTC_NO_THREAD_SAFETY_ANALYSIS is an attribute on functions or methods, which turns off thread safety
checking for that method. It provides an escape hatch for functions which are either
(1) deliberately thread-unsafe, or
(2) are thread-safe, but too complicated for the analysis to understand.
Bug: webrtc:9202
Change-Id: Ie332bca7eb7eb535ed965de5ddc42872c4f30602
Reviewed-on: https://webrtc-review.googlesource.com/76562
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23221}
Changed minimum bit rate threshold formula to raise the minimum bit rate
at which 720p video is presented in VP9 SVC to ensure that the video
quality for VP9 SVC is the same or better than VP8 SIM. The minimum bit
rate threshold values for lower resolutions remain largely unchanged.
Also changed maximum bit rate threshold formula to lower the maximum
bit rate for low resolutions (e.g., 180p) in order to ensure higher
frame rates when downlink bit rates are very low (e.g., < 100 kbps).
Bug: webrtc:9242
Change-Id: I8f9c76c9188b98f3fd40a608551b576b0c3b8f34
Reviewed-on: https://webrtc-review.googlesource.com/75244
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23218}
The previous effort of building WebRTC without SW codecs stopped when
libjingle_peerconnection was possible to build. In order to make the
group("default") target build, this basically updates a bunch of
tests to explicitly depend on the built-in software video codecs.
Bug: webrtc:7925
Change-Id: I2715414770c197fca01cb8dbde173a21f4434500
Reviewed-on: https://webrtc-review.googlesource.com/70503
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23216}
Mini-change: add 'webrtc' namespace. The template class AudioFrameView
got declared in the global namespace by mistake. (My fault). Now
fixing.
Bug: webrtc:9262.
Change-Id: I6f2b4ab1ccdb223505e7181b8e6f12f5f23b3684
Reviewed-on: https://webrtc-review.googlesource.com/76140
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23215}
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.
The unit tests are re-implemented as XCTests.
(was: https://webrtc-review.googlesource.com/c/src/+/67300)
Bug: webrtc:9120
Change-Id: I07340505137b16c2dd487569ad0112f984557bba
Reviewed-on: https://webrtc-review.googlesource.com/75125
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23208}
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.
Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
This CL defines SpectralFeaturesExtractor which is responsible for
computing the spectral features used as input for the RNN.
Bug: webrtc:9076
Change-Id: I5e1396b89eca9c13bb268e8419a16436a9c3450f
Reviewed-on: https://webrtc-review.googlesource.com/73760
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23206}
This removes second reference for frame 3 in GOF predefined for 3
temporal layers since encoder never use that reference.
Bug: webrtc:9245
Change-Id: I6fbdbe7d3c753dda7fbcfcbd05f3530f70f80728
Reviewed-on: https://webrtc-review.googlesource.com/74705
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23193}
This CL adds robustness to avoid the AEC3 transparent mode to be
incorrectly activated when
-there is strong near-end noise
-there is only low-level nearend activity.
Bug: webrtc:9256,chromium:841193
Change-Id: I26c2759d163914eb85dc3d863da8acbf28cbb88d
Reviewed-on: https://webrtc-review.googlesource.com/75511
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23191}
This CL corrects the usage of the estimated echo path gain to not be
hardcoded to 1. In order to retain the tuned behavior, the CL for now
maintains the former behavior in the code.
Bug: webrtc:9255,chromium:851187
Change-Id: I7f91c72e476680a8a854c22b74b1771fae446110
Reviewed-on: https://webrtc-review.googlesource.com/75510
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23190}
This CL ensures that the external audio buffer delay is correctly used
by removing the applied headroom and avoiding that the delay estimation
feedback fromt the echo remover overrules the external delay
information.
Bug: webrtc:9241,chromium:839860
Change-Id: I53cc78ace34a71994ab24a3b552f29979e2aae78
Reviewed-on: https://webrtc-review.googlesource.com/75513
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23189}
This prepares for allowing injection of a network controller.
Bug: webrtc:9155
Change-Id: I5624f47738db9c5cd4750eac76cb6289e06a7aa3
Reviewed-on: https://webrtc-review.googlesource.com/73100
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23188}
This CL is part of the effort to remove warning suppression flags from
the WebRTC build.
Bug: webrtc:9251
Change-Id: I45ece25e897a14a6d4ce8a90ba59688f8fc6fe32
Reviewed-on: https://webrtc-review.googlesource.com/75503
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23187}
Before this CL, there would be an out-of-bounds write in the ByteBuffer
copying when a decoded frame had height != sliceHeight.
Bug: webrtc:9194
Change-Id: Ibb80e5555e8f00d9e1fd4cb8a73f5e4ccd5a0b81
Tested: 640x360 loopback with eglContext == null in AppRTCMobile on Pixel.
Reviewed-on: https://webrtc-review.googlesource.com/74120
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23184}
This change adds flags and default values for two more RTP header
extensions: VideoContentType and VideoTiming.
This will silence a number of annoying warnings when running with
application logs.
Bug: none
Change-Id: I9bb01ea2519813d3c47553ecff384141fbede23e
Reviewed-on: https://webrtc-review.googlesource.com/75300
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23178}
This removes the redundant type and replaces all usages. A slight change
in behavior is that we no longer get nanosecond resolution. This should
not matter since no current code requires nanosecond resolution.
Bug: webrtc:9155
Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2
Reviewed-on: https://webrtc-review.googlesource.com/71163
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23174}
During pitch search in the RNN VAD, we calculate auto
correlation. Before this CL, we computed kNumInvertedLags12kHz=147 dot
products of vectors with kBufSize12kHz-kMaxPitch12kHz=240
elements. This was the most time consuming step of the new VAD.
This CL makes the computation happen in frequency domain. Profiling
shows a 3x speed increase. In future, we can try using a more efficient
FFT and to reduce the FFT length to some of e.g. 400, 405, 432.
# For minimal Clang plugin check change.
TBR: kwiberg@webrtc.org
Bug: webrtc:9076
Change-Id: I688251a415869d53175a37f390f441d4e035d954
Reviewed-on: https://webrtc-review.googlesource.com/73366
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23171}
This CL adds helper functions to be used for the spectral features
computation. Namely, it includes the following:
- band boundaries (frequency to FFT coeffcient index)
- band energy coefficients
- log band energy coefficients
- fixed size DCT table and computation
Bug: webrtc:9076
Change-Id: I03a8799b226d986bc1e37cefd0c3039f94b5592a
Reviewed-on: https://webrtc-review.googlesource.com/73687
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23170}
This prepares for being able to inject network congestion controllers.
And makes it easier to use the units in other parts of the code.
Bug: webrtc:9155
Change-Id: Ib8f9c1c97b06d791a01c3376046933d576ae46f9
Reviewed-on: https://webrtc-review.googlesource.com/70201
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23168}
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.
The unit tests are re-implemented as XCTests.
Bug: webrtc:9120
Change-Id: Ie60cafae796efbd7966d21ff6877c92cbe850fb7
Reviewed-on: https://webrtc-review.googlesource.com/67300
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23163}
This makes the dependency graph simpler and prepares for moving the
unit classes to api/.
Bug: webrtc:9155
Change-Id: I1b36d5e05f75d70ba8951e880d76359f896f7741
Reviewed-on: https://webrtc-review.googlesource.com/74920
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23161}
Always having the latest iosurface before invalidating a region.
Otherwise if CaptureFrame() happens in between, the capture result
may not be fully refreshed. Also we can't add lock since it will
impact performance.
Bug: webrtc:8652
Change-Id: Ib23105b16065018c691685083b76a771ce8771d3
Reviewed-on: https://webrtc-review.googlesource.com/74643
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23154}
BandAnalysisFft class that wraps the FFT library, makes it easy to change
FFT library, applies windowing function and owns the FFT input buffer.
Bug: webrtc:9076
Change-Id: I9e7ed587ae263b906e04a66bf8c06eaae64daf19
Reviewed-on: https://webrtc-review.googlesource.com/72900
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23150}
This simplifies the code and removes the need for a lot of bookkeeping
variables.
Bug: webrtc:9232
Change-Id: I0c9a4b0741ed5353caa22ba5acdcb166357441f2
Reviewed-on: https://webrtc-review.googlesource.com/74240
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23149}
This CL softens the effect of the AEC3 transparent mode to also handle
headsets that leak low-level echoes in a nonlinear way.
This is handled by reintroducing the limit in the echo path gain for the
nonlinear mode. Due to recent improvements in echo suppressor behavior
this is now possible to do with a limited impact on the near-end speech.
Bug: webrtc:9246,chromium:840347
Change-Id: I0ca5157160d1884ba93b962323b56016756986d3
Reviewed-on: https://webrtc-review.googlesource.com/74703
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23145}
Adding a build target for the bi-qaud filter to make it available for
the RNN VAD of AGC2. Also adding a unit test to test the computation
both in-place and not in-place while comparing the produced output to
that of scipy.signal.
Bug: webrtc:9076
Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772
Reviewed-on: https://webrtc-review.googlesource.com/74220
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23141}
