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: I46c09900246f75ca5285aeb38f7b8b295784ffac
Reviewed-on: https://webrtc-review.googlesource.com/76741
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23238}
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}
Vp9 encoder supports several inter-layer prediction modes. This adds
possibility to control and test them in video/ss/sv loopback.
Filtering of sent packets has been modified. In addition to high
spatial and temporal layers it now filters out packets of low spatial
layers where non_ref_for_inter_layer_pred bit is set to true.
Bug: none
Change-Id: I17b1ee8f1ac1d70a6914eb86d153790ef2da9679
Reviewed-on: https://webrtc-review.googlesource.com/76540
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23233}
Start using JniIntWrapper from Chromium instead of bypassing
it in jni_generator_helper.h.
Bug: webrtc:8278
Change-Id: I20313e1e610b05f79c210e823ab50cfb2073674e
Reviewed-on: https://webrtc-review.googlesource.com/74841
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23230}
To aid deletion of deprecated methods in external child classes
of PeerConnectionFactoryInterface.
Bug: webrtc:9239
Change-Id: Idbc1c31285bceeb172a0b5bdb2b106f0c449dcb9
Reviewed-on: https://webrtc-review.googlesource.com/76623
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23229}
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}
This unconditionally limits usage of VP9 SVC inter-layer prediction to
frames of key picture.
Ideally we would like to let an application to configure SVC options.
But currently there is no API for this.
Bug: none
Change-Id: I21a84dafc946be122514d5b6bf327b65251f1115
Reviewed-on: https://webrtc-review.googlesource.com/76640
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23224}
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}
This CL allows to reduce the code required to create fixed-size ArrayView
objects for std::array instances. Instead of passing .data() and .size(),
it is now sufficient to pass the std::array instance. When instancing an
array view with variable size, a different ctor is called.
Bug: webrtc:9076
Change-Id: I4fe133b27cd12827ed0206d40184279fc3a196f5
Reviewed-on: https://webrtc-review.googlesource.com/76160
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23220}
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}
This should fix our python presubmit tests.
Also make Artem and Patrik own the deps file.
Bug: webrtc:9264
Change-Id: I430c56e0b12c4feac7d7cbf766eea9e304cd9a20
Reviewed-on: https://webrtc-review.googlesource.com/76421
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23217}
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}
It is stored in a text file as a stream of 'r' and 'c' characters - render and capture.
This is the format output by APM with apm_debug_dump on, and it is readable by audioproc_f.
Bug: webrtc:9252
Change-Id: I01e9e104ed7e3fb45e623730343a0c2addc81d1b
Reviewed-on: https://webrtc-review.googlesource.com/75502
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23213}
Fix assert that breaks on certain bots.
Bug: webrtc:9228
Change-Id: Ie28ba4a27a4ca07a6877373e41a432ee67eccb1e
Reviewed-on: https://webrtc-review.googlesource.com/76441
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23212}
Previously, constructing a PeerConnection or WebRtcVideoEngine with
fake encoder/decoder factories would result in the real, built-in factories
also being used. In https://webrtc-review.googlesource.com/c/src/+/71162, this
changed, so to temporarily allow tests to continue working exactly the same as
before, the fake factories started encapsulating the real factories. This CL
removes that behavior and updates the tests accordingly.
Bug: webrtc:9228
Change-Id: Ida14a1e3f5f5a0e2f03100b7895b3b1bdf0a0a42
Reviewed-on: https://webrtc-review.googlesource.com/75260
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23209}
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}
GetAdapterTypeFromName determines the adapter type of a network
interface based on the string matching of the interface name. It however
does not have an entry to map the well-known "eth" name to the common
Ethernet type. This introduces subtle bugs when GetAdapterTypeFromName
is used as the only method to determine a network type and Ethernet is
thus identified as an unknown network, which affects the network
filtering and network path selection that rely on the network type.
Bug: webrtc:9235
Change-Id: Ifc3269d191382f3b3a041de1c9755c09994b31b2
Reviewed-on: https://webrtc-review.googlesource.com/74263
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23195}
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}
The fuzzer is very simple. It only considers the default encoder
configuration at this point.
Bug: chromium:826914
Change-Id: Ifa248a1dba80efb231807750e40082ec5580636a
Reviewed-on: https://webrtc-review.googlesource.com/75261
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23192}
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}
