doc: Add some documentation about the release process

Will allow us to build the libs as part of gst-build as subprojects.

.gitignore

@@ -1,30 +1,10 @@
 *.o
-*.lo
-*.la
 *.pc
 .*.swp
 *~
 .deps*
+.dirstamp
 .libs*
-Makefile
-Makefile.in
-aclocal.m4
-autom4te.cache
-compile
-config.guess
-config.h
-config.h.in
-config.log
-config.rpath
-config.status
-config.sub
-configure
+build/
 depcomp
-install-sh
-libltdl
-libtool
-ltmain.sh
-missing
-mkinstalldirs
-stamp-*
-
+install/

.gitlab-ci.yml

@@ -0,0 +1,98 @@
+# The build has two stages. The 'container' stage is used to build a Docker
+# container and push it to the project's container registry on fd.o GitLab.
+# This step is only run when the tag for the container changes, else it is
+# effectively a no-op. All of this infrastructure is inherited from the
+# wayland/ci-templates repository which is the recommended way to set up CI
+# infrastructure on fd.o GitLab.
+#
+# Once the container stage is done, we move on to the 'build' stage where we
+# run an autotools and meson build in parallel. Currently, tests are also run
+# as part of the build stage as there doesn't seem to be significant value to
+# splitting the stages at the moment.
+
+stages:
+  - container
+  - build
+
+variables:
+  # Update this tag when you want to trigger a rebuild the container in which
+  # CI runs, for example when adding new packages to FDO_DISTRIBUTION_PACKAGES.
+  # The tag is an arbitrary string that identifies the exact container
+  # contents.
+  BASE_TAG: '2020-11-27.1'
+  FDO_DISTRIBUTION_VERSION: '20.10'
+  FDO_UPSTREAM_REPO: 'pulseaudio/webrtc-audio-processing'
+
+include:
+  # We pull templates from master to avoid the overhead of periodically
+  # scanning for changes upstream. This does means builds might occasionally
+  # break due to upstream changing things, so if you see unexpected build
+  # failures, this might be one cause.
+  - project: 'freedesktop/ci-templates'
+    ref: 'master'
+    file: '/templates/ubuntu.yml'
+
+# Common container build template
+.ubuntu-container-build:
+  variables:
+    GIT_STRATEGY: none # no need to pull the whole tree for rebuilding the image
+
+    # Remember to update FDO_DISTRIBUTION_TAG when modifying this package list!
+    # Otherwise the changes won't have effect since an old container image will
+    # be used.
+    FDO_DISTRIBUTION_PACKAGES: >-
+      g++
+      gcc
+      libabsl-dev
+      meson
+      ninja-build
+      python3-setuptools
+
+# Used to extend both container and build jobs
+.ubuntu-x86_64:
+  variables:
+    FDO_DISTRIBUTION_TAG: "x86_64-$BASE_TAG"
+
+# Used to extend both container and build jobs
+.ubuntu-aarch64:
+  tags:
+    - aarch64
+  variables:
+    FDO_DISTRIBUTION_TAG: "aarch64-$BASE_TAG"
+
+build-container-x86_64:
+  extends:
+    - .fdo.container-build@ubuntu@x86_64
+    - .ubuntu-container-build
+    - .ubuntu-x86_64
+  stage: container
+
+build-container-aarch64:
+  extends:
+    - .fdo.container-build@ubuntu@aarch64
+    - .ubuntu-container-build
+    - .ubuntu-aarch64
+  stage: container
+
+# Common build template
+.build:
+  stage: build
+  extends:
+    - .fdo.distribution-image@ubuntu
+  script:
+    - meson build
+    - cd build
+    - ninja
+  artifacts:
+    paths:
+      - build/
+
+build-x86_64:
+  extends:
+    - .build
+    - .ubuntu-x86_64
+
+build-aarch64:
+  extends:
+    - .build
+    - .ubuntu-aarch64

Makefile.am

@@ -1,6 +0,0 @@
-SUBDIRS = src
-
-pkgconfigdir = $(libdir)/pkgconfig
-pkgconfig_DATA = webrtc-audio-processing.pc
-
-EXTRA_DIST = PATENTS

NEWS

@@ -1,3 +1,53 @@
+Release 0.3
+-----------
+
+Minor build fixes.
+
+
+Release 0.2
+-----------
+
+Updated AudioProcessing code to be more current.
+
+Contains API breaking changes.
+
+Upstream changes include:
+
+  * Rewritten AGC and voice activity detection
+  * Intelligibility enhancer
+  * Extended AEC filter
+  * Beamformer
+  * Transient suppressor
+  * ARM, NEON and MIPS optimisations (MIPS optimisations are not hooked up)
+
+API changes:
+
+  * We no longer include a top-level audio_processing.h. The webrtc tree format
+    is used, so use webrtc/modules/audio_processing/include/audio_processing.h
+  * The top-level module_common_types.h has also been moved to
+    webrtc/modules/interface/module_common_types.h
+  * C++11 support is now required while compiling client code
+  * AudioProcessing::Create() does not take any arguments any more
+  * AudioProcessing::Destroy() is gone, use standard C++ "delete" instead
+  * Stream parameters are now configured via StreamConfig and ProcessingConfig
+    rather than set_sample_rate(), set_num_channels(), etc.
+  * AudioFrame field names have changed
+  * Use config API for newer audio processing options
+  * Use ProcessReverseStream() instead of AnalyzeReverseStream(), particularly
+    when using the intelligibility enhancer
+  * GainControl::set_analog_level_limits() is broken. The AGC implementation
+    hard codes 0-255 as the volume range
+
+Other notes:
+
+  * The new audio processing parameters are not all tested, and a few are not
+    enabled upstream (in Chromium) either
+  * The rewritten AGC appears to be less sensitive, and it might make sense to
+    initialise the capture volume to something reasonable (33% or 50%, for
+    example) to make sure there is sufficient energy in the stream to trigger
+    the AGC mechanism
+
+
 Release 0.1
 -----------
 

README

@@ -1,40 +1 @@
-About
-=====
-
-This is meant to be a more Linux packaging friendly copy of the AudioProcessing
-module from the WebRTC[1] project. The ideal case is that we make no changes to
-the code to make tracking upstream code easy.
-
-This package currently only includes the AudioProcessing bits, but I am very
-open to collaborating with other projects that wish to distribute other bits of
-the code and hopefully eventually have a single point of packaging all the
-WebRTC code to help people reuse the code and avoid keeping private copies in
-several different projects.
-
-[1] http://code.google.com/p/webrtc/
-
-Feedback
-========
-
-Patches, suggestions welcome. You can send them to the PulseAudio mailing
-list[2] or to me at the address below.
-
--- Arun Raghavan <arun.raghavan@collabora.co.uk>
-
-[2] http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
-
-Notes
-=====
-
-Assembling some quick notes on maintaining this tree vs. the original WebRTC
-project source code.
-
-1. Running meld on a pristine tree's src/ vs. the same directory in the WebRTC
-   code should produce a fairly easy-to-parse set of differences that can be
-   merged. I've kept the test code out for now, but this might get merged in
-   the future. The .gyp/.gypi files are included to make tracking addition and
-   removal of files easy to track.
-
-2. Be sure to keep the keep the comment on top of the version field in
-   configure.ac up-to-date so that it's easy to track what changes went in at
-   what point.
+See README.md

README.md

@@ -0,0 +1,35 @@
+About
+=====
+
+This is meant to be a more Linux packaging friendly copy of the AudioProcessing
+module from the WebRTC[1] project. The ideal case is that we make no changes to
+the code to make tracking upstream code easy.
+
+This package currently only includes the AudioProcessing bits, but I am very
+open to collaborating with other projects that wish to distribute other bits of
+the code and hopefully eventually have a single point of packaging all the
+WebRTC code to help people reuse the code and avoid keeping private copies in
+several different projects.
+
+[1] https://webrtc.googlesource.com/src
+
+Feedback
+========
+
+Patches, suggestions welcome. You can send them to the PulseAudio mailing
+list[2] or to me at the address below.
+
+-- Arun Raghavan <mail@arunraghavan.net>
+
+[2] http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
+
+Notes
+====
+
+1. Some files need to be patch to avoid pulling in the gtest framework. This
+   should ideally be pushed upstream in some way so we're able to just pull
+   in what we need without changing anything.
+
+2. It might be nice to try LTO on the library. We build a lot of code as part
+   of the main AudioProcessing module deps, and it's possible that this could
+   provide significant space savings.

RELEASING.md

@@ -0,0 +1,61 @@
+# Release process
+
+## Update the code
+
+Follow the instructions in `UPDATING.md` to update the code.
+
+## Update the package version
+
+If there is no API breakage, update the minor version (X.y -> X.y+1). If there
+is API breakage, update the major version (X.y -> X+1.0).
+
+## Make sure builds are successful on all platforms
+
+There is CI for `x86_64` and `aarch64` builds, but 32-bit ARM and MIPS builds
+need manual verfification (or testing downstream).
+
+## Tag the release
+
+Tag the release with:
+
+```sh
+git tag -s -m 'WebRTC AudioProcessing v<X.y>' v<X.y>
+```
+
+## Make a tarball
+
+```sh
+git archive --format 'tar.gz' \
+  --prefix webrtc-audio-processing-X.y -9 vX.y \
+  > webrtc-audio-processing-X.y.tar.gz
+```
+
+## Checksum the tarball
+
+```sh
+sha256sum webrtc-audio-processing-X.y.tar.gz \
+  > webrtc-audio-processing-X.y.tar.gz.sha256
+```
+
+## Publish the files
+
+```sh
+scp webrtc-audio-processing-*.tar.* \
+  annarchy.freedesktop.org:/srv/www.freedesktop.org/www/software/pulseaudio/webrtc-audio-processing/
+```
+
+## Push the tag
+
+```sh
+git push origin master
+git push origin vX.y
+```
+
+## Update the website
+
+This is currently an embarrassing manual process.
+
+## Send out a release announcement
+
+This goes to the `pulseaudio-discuss` and `gstreamer-devel` mailing lists, and
+possibly `discuss-webrtc` if it seems relevant.

UPDATING.md

@@ -0,0 +1,66 @@
+Updating
+=====
+
+Assembling some quick notes on maintaining this tree vs. the upstream WebRTC
+project source code.
+
+1. The code is currently synced agains whatever revision of the upstream
+   webrtc git repository Chromium uses.
+
+2. Instructions on checking out the Chromium tree are on the
+   [WebRTC repo][get-webrtc]. As a shortcut, you can look at the DEPS file
+   in the Chromium tree for the current webrtc version being used, and then
+   just use that commit hash with the webrtc tree.
+
+3. [Meld][meld] is a great tool for diffing two directories. Start by running
+   it on ```webrtc-audio-processing/webrtc``` and
+   ```chromium/third_party/webrtc```.
+
+   * For each directory in the ```webrtc-audio-processing``` tree, go over the
+     corresponding code in the ```chromium``` tree.
+
+   * Examine changed files, and pick any new changes. A small number of files
+     in the ```webrtc-audio-processing``` tree have been changed by hand, make
+     sure that those are not overwritten.
+
+   * unittest files have been left out since they are not built or used.
+
+   * BUILD.gn files have been copied to keep track of changes to the build
+     system upstreama.
+
+   * Arch-specific files usually have special handling in the corresponding
+     meson.build.
+
+4. Once everything has been copied and updated, everything needs to be built.
+   Missing dependencies (files that were not copied, or new modules that are
+   being depended on) will first turn up here.
+
+   * Copy new deps as needed, leaving out testing-only dependencies insofar as
+     this is possible.
+
+5. ```webrtc/modules/audio_processing/include/audio_processing.h``` is the main
+   include file, so look for API changes here.
+
+   * The current policy is that we mirror upstream API as-is.
+
+   * Update soversion in meson.build with the appropriate version info  based on how the
+     code has changed. Details on how to do this are included in the
+     [libtool documentation][libtool-version-info].
+
+5. Build PulseAudio (and/or any other dependent projects) against the new code.
+   The easy way to do this is via a prefixed install.
+
+   * Configure webrtc-audio-processing with
+     ```meson build -D prefix=$(pwd)/install```, then do a ```ninja -C build/ install```
+
+   * Configure PulseAudio with
+     ```meson build -D pkg_config_path=/path/to/webrtc-audio-processing/install/lib64/pkgconfig/```, which will cause the
+     build to pick up the prefixed install. Then do a ```ninja -C build```, run the built
+     PulseAudio, and load ```module-echo-cancel``` to make sure it loads fine.
+
+   * Run some test streams through the canceller to make sure it is working
+     fine.
+
+[get-webrtc]: https://webrtc.googlesource.com/src/
+[meld]: http://meldmerge.org/
+[libtool-version-info]: https://www.gnu.org/software/libtool/manual/html_node/Updating-version-info.html

autogen.sh

@@ -1,6 +0,0 @@
-#!/bin/sh
-libtoolize
-aclocal
-automake --add-missing --copy
-autoconf
-./configure ${@}

configure.ac

@@ -1,49 +0,0 @@
-# Revision changelog (version - date, svn rev. from upstream that was merged)
-#    0.1 - 21 Oct 2011, r789
-AC_INIT([webrtc-audio-processing], [0.1])
-AM_INIT_AUTOMAKE([dist-xz tar-ustar])
-
-AC_SUBST(LIBWEBRTC_AUDIO_PROCESSING_VERSION_INFO, [0:0:0])
-
-AM_SILENT_RULES([yes])
-
-AC_PROG_CC
-AC_PROG_CXX
-AC_PROG_LIBTOOL
-AC_PROG_INSTALL
-
-AC_LANG_C
-AC_LANG_CPLUSPLUS
-
-AC_ARG_WITH([ns-mode],
-	    AS_HELP_STRING([--with-ns-mode=float|fixed], [Noise suppresion mode to use. Default is float]))
-
-AS_CASE(["x${with_ns_mode}"],
-	["fixed"], [NS_FIXED=1],
-	["float"], [NS_FIXED=0],
-	[NS_FIXED=0])
-AM_CONDITIONAL(NS_FIXED, [test "x${NS_FIXED}" = "x1"])
-
-COMMON_CFLAGS="-DNDEBUG -I\$(srcdir)/interface -I\$(srcdir)/main/interface -I\$(top_srcdir)/src -I\$(top_srcdir)/src/modules/interface"
-COMMON_CXXFLAGS="-DNDEBUG -I\$(srcdir)/interface -I\$(srcdir)/main/interface -I\$(top_srcdir)/src -I\$(top_srcdir)/src/modules/interface"
-AC_SUBST([COMMON_CFLAGS])
-AC_SUBST([COMMON_CXXFLAGS])
-
-AC_CONFIG_FILES([
-webrtc-audio-processing.pc
-Makefile
-src/Makefile
-src/common_audio/Makefile
-src/common_audio/signal_processing_library/Makefile
-src/common_audio/vad/Makefile
-src/system_wrappers/Makefile
-src/modules/Makefile
-src/modules/audio_processing/Makefile
-src/modules/audio_processing/utility/Makefile
-src/modules/audio_processing/ns/Makefile
-src/modules/audio_processing/aec/Makefile
-src/modules/audio_processing/aecm/Makefile
-src/modules/audio_processing/agc/Makefile
-])
-
-AC_OUTPUT

meson.build

@@ -0,0 +1,184 @@
+project('webrtc-audio-processing-1', 'c', 'cpp',
+  version : '1.0',
+  meson_version : '>= 0.54',
+  default_options : [ 'warning_level=1',
+                      'buildtype=debugoptimized',
+                      'c_std=c11',
+                      'cpp_std=c++14',
+                    ]
+)
+
+version_split = meson.project_version().split('.')
+# This will be incremented each time a breaking API change occurs
+major_version = version_split[0]
+# This will be incremented when there are backwards-compatible changes
+minor_version = version_split[1]
+
+# We maintain per-package versions to not have to break API for one if only the
+# other has breaking changes
+
+apm_major_version = major_version
+apm_minor_version = minor_version
+apm_version = apm_major_version + '.' + apm_minor_version
+apm_project_name = 'webrtc-audio-processing-' + apm_major_version
+
+ac_major_version = major_version
+ac_minor_version = minor_version
+ac_version = ac_major_version + '.' + ac_minor_version
+ac_project_name = 'webrtc-audio-coding-' + ac_major_version
+
+include_subdir = meson.project_name()
+
+cc = meson.get_compiler('c')
+cpp = meson.get_compiler('cpp')
+
+host_system = host_machine.system()
+
+platform_cflags = []
+os_cflags = []
+os_deps = []
+have_posix = false
+have_win = false
+
+absl_dep = [
+  cpp.find_library('absl_base'),
+  cpp.find_library('absl_bad_optional_access'),
+  cpp.find_library('absl_flags_internal'),
+  cpp.find_library('absl_flags_marshalling'),
+  cpp.find_library('absl_flags_parse'),
+  cpp.find_library('absl_flags_registry'),
+  cpp.find_library('absl_flags_usage_internal'),
+  cpp.find_library('absl_raw_logging_internal'),
+  cpp.find_library('absl_strings'),
+  cpp.find_library('absl_synchronization'),
+  cpp.find_library('absl_throw_delegate'),
+]
+
+if ['darwin', 'ios'].contains(host_system)
+  os_cflags = ['-DWEBRTC_MAC']
+  if host_system == 'ios'
+    os_cflags += ['-DWEBRTC_IOS']
+  endif
+  have_posix = true
+elif host_system == 'android'
+  os_cflags += ['-DWEBRTC_ANDROID', '-DWEBRTC_LINUX']
+  os_deps += [cc.find_library('log')]
+  os_deps += [dependency('gnustl', required : get_option('gnustl'))]
+  have_posix = true
+elif host_system == 'linux'
+  os_cflags += ['-DWEBRTC_LINUX', '-DWEBRTC_THREAD_RR']
+  os_deps += [cc.find_library('rt', required : false)]
+  os_deps += [dependency('threads')]
+  have_posix = true
+elif host_system == 'windows'
+  platform_cflags += ['-DWEBRTC_WIN', '-D_WIN32', '-U__STRICT_ANSI__']
+  os_deps += [cc.find_library('winmm')]
+  have_win = true
+endif
+
+if have_posix
+  platform_cflags += ['-DWEBRTC_POSIX']
+endif
+
+arch_cflags = []
+have_arm = false
+have_armv7 = false
+have_neon = false
+have_mips = false
+have_mips64 = false
+have_x86 = false
+have_avx2 = false
+if ['arm', 'armv7'].contains(host_machine.cpu_family())
+  if cc.compiles('''#ifdef __ARM_ARCH_ISA_ARM
+#error no arm arch
+#endif''')
+    have_arm = true
+    arch_cflags += ['-DWEBRTC_ARCH_ARM']
+  endif
+  if cc.compiles('''#ifndef __ARM_ARCH_7A__
+#error no armv7 arch
+#endif''')
+    have_armv7 = true
+    arch_cflags += ['-DWEBRTC_ARCH_ARM_V7']
+  endif
+endif
+if cc.compiles('''#ifndef __aarch64__
+#error no aarch64 arch
+#endif''')
+  have_neon = true
+  arch_cflags += ['-DWEBRTC_ARCH_ARM64', '-DWEBRTC_HAS_NEON']
+endif
+if ['mips', 'mips64'].contains(host_machine.cpu_family())
+  have_mips = true
+  arch_cflags += ['WEBRTC_ARCH_MIPS_FAMILY']
+endif
+if host_machine.cpu_family() == 'mips64'
+  have_mips64 = true
+endif
+if ['x86', 'x86_64'].contains(host_machine.cpu_family())
+  have_x86 = true
+  # This is unconditionally enabled for now, actual usage is determined by
+  # runtime CPU detection, so we're just assuming the compiler supports avx2
+  have_avx2 = true
+  arch_cflags += ['-DWEBRTC_ENABLE_AVX2']
+endif
+
+neon_opt = get_option('neon')
+if neon_opt != 'no'
+  if neon_opt != 'runtime'
+    if cc.compiles('#include <arm_neon.h>', args : '-mfpu=neon')
+      arch_cflags += ['-mfpu=neon', '-DWEBRTC_HAS_NEON']
+      have_neon = true
+    endif
+  else
+    neon_opt += ['-DWEBRTC_DETECT_NEON', '-mfpu=neon']
+    have_neon = true
+  endif
+endif
+
+common_cflags = [
+  '-DWEBRTC_LIBRARY_IMPL',
+  '-DWEBRTC_ENABLE_SYMBOL_EXPORT',
+  '-DNDEBUG'
+  ] + platform_cflags + os_cflags + arch_cflags
+common_cxxflags = common_cflags
+common_deps = os_deps + [absl_dep]
+webrtc_inc = include_directories('.')
+
+subdir('webrtc')
+
+pkgconfig = import('pkgconfig')
+
+pkgconfig.generate(
+    name: apm_project_name,
+    description: 'WebRTC Audio Processing library',
+    version: apm_major_version + '.' + apm_minor_version,
+    filebase: apm_project_name,
+    subdirs: include_subdir,
+    extra_cflags: [
+      '-DWEBRTC_LIBRARY_IMPL',
+    ] + platform_cflags,
+    libraries: libwebrtc_audio_processing,
+)
+
+audio_processing_dep = declare_dependency(link_with : libwebrtc_audio_processing,
+    include_directories : [webrtc_inc])
+
+meson.override_dependency(apm_project_name, audio_processing_dep)
+
+pkgconfig.generate(
+    name: ac_project_name,
+    description: 'WebRTC Audio Coding library',
+    version: ac_major_version + '.' + ac_minor_version,
+    filebase: ac_project_name,
+    subdirs: include_subdir,
+    extra_cflags: [
+      '-DWEBRTC_LIBRARY_IMPL',
+    ] + platform_cflags,
+    libraries: libwebrtc_audio_coding,
+)
+
+audio_coding_dep = declare_dependency(link_with : libwebrtc_audio_coding,
+    include_directories : [webrtc_inc])
+
+meson.override_dependency(ac_project_name, audio_coding_dep)

meson_options.txt

@@ -0,0 +1,6 @@
+option('gnustl', type: 'feature',
+       value: 'auto',
+       description: 'Use gnustl for a c++ library implementation (only used on Android)')
+option('neon', type: 'combo',
+       choices: ['no', 'yes', 'auto', 'runtime'],
+       description: '')

src/Makefile.am

@@ -1 +0,0 @@
-SUBDIRS = common_audio system_wrappers modules

src/common_audio/Makefile.am

@@ -1 +0,0 @@
-SUBDIRS = signal_processing_library vad

src/common_audio/signal_processing_library/Makefile.am

@@ -1,44 +0,0 @@
-noinst_LTLIBRARIES = libspl.la
-
-libspl_la_SOURCES =  main/interface/signal_processing_library.h \
-		     main/interface/spl_inl.h \
-		     main/source/auto_corr_to_refl_coef.c \
-		     main/source/auto_correlation.c \
-		     main/source/complex_fft.c \
-		     main/source/complex_ifft.c \
-		     main/source/complex_bit_reverse.c \
-		     main/source/copy_set_operations.c \
-		     main/source/cos_table.c \
-		     main/source/cross_correlation.c \
-		     main/source/division_operations.c \
-		     main/source/dot_product_with_scale.c \
-		     main/source/downsample_fast.c \
-		     main/source/energy.c \
-		     main/source/filter_ar.c \
-		     main/source/filter_ar_fast_q12.c \
-		     main/source/filter_ma_fast_q12.c \
-		     main/source/get_hanning_window.c \
-		     main/source/get_scaling_square.c \
-		     main/source/hanning_table.c \
-		     main/source/ilbc_specific_functions.c \
-		     main/source/levinson_durbin.c \
-		     main/source/lpc_to_refl_coef.c \
-		     main/source/min_max_operations.c \
-		     main/source/randn_table.c \
-		     main/source/randomization_functions.c \
-		     main/source/refl_coef_to_lpc.c \
-		     main/source/resample.c \
-		     main/source/resample_48khz.c \
-		     main/source/resample_by_2.c \
-		     main/source/resample_by_2_internal.c \
-		     main/source/resample_by_2_internal.h \
-		     main/source/resample_fractional.c \
-		     main/source/sin_table.c \
-		     main/source/sin_table_1024.c \
-		     main/source/spl_sqrt.c \
-		     main/source/spl_sqrt_floor.c \
-		     main/source/spl_version.c \
-		     main/source/splitting_filter.c \
-		     main/source/sqrt_of_one_minus_x_squared.c \
-		     main/source/vector_scaling_operations.c
-libspl_la_CFLAGS = $(AM_CFLAGS) $(COMMON_CFLAGS)

src/common_audio/signal_processing_library/main/interface/spl_inl.h

@@ -1,159 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-// This header file includes the inline functions in
-// the fix point signal processing library.
-
-#ifndef WEBRTC_SPL_SPL_INL_H_
-#define WEBRTC_SPL_SPL_INL_H_
-
-#ifdef WEBRTC_ARCH_ARM_V7A
-#include "spl_inl_armv7.h"
-#else
-
-static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
-  WebRtc_Word16 out16 = (WebRtc_Word16) value32;
-
-  if (value32 > 32767)
-    out16 = 32767;
-  else if (value32 < -32768)
-    out16 = -32768;
-
-  return out16;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
-                                                  WebRtc_Word16 b) {
-  return WebRtcSpl_SatW32ToW16((WebRtc_Word32) a + (WebRtc_Word32) b);
-}
-
-static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
-                                                  WebRtc_Word32 l_var2) {
-  WebRtc_Word32 l_sum;
-
-  // perform long addition
-  l_sum = l_var1 + l_var2;
-
-  // check for under or overflow
-  if (WEBRTC_SPL_IS_NEG(l_var1)) {
-    if (WEBRTC_SPL_IS_NEG(l_var2) && !WEBRTC_SPL_IS_NEG(l_sum)) {
-        l_sum = (WebRtc_Word32)0x80000000;
-    }
-  } else {
-    if (!WEBRTC_SPL_IS_NEG(l_var2) && WEBRTC_SPL_IS_NEG(l_sum)) {
-        l_sum = (WebRtc_Word32)0x7FFFFFFF;
-    }
-  }
-
-  return l_sum;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
-                                                  WebRtc_Word16 var2) {
-  return WebRtcSpl_SatW32ToW16((WebRtc_Word32) var1 - (WebRtc_Word32) var2);
-}
-
-static __inline WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
-                                                  WebRtc_Word32 l_var2) {
-  WebRtc_Word32 l_diff;
-
-  // perform subtraction
-  l_diff = l_var1 - l_var2;
-
-  // check for underflow
-  if ((l_var1 < 0) && (l_var2 > 0) && (l_diff > 0))
-    l_diff = (WebRtc_Word32)0x80000000;
-  // check for overflow
-  if ((l_var1 > 0) && (l_var2 < 0) && (l_diff < 0))
-    l_diff = (WebRtc_Word32)0x7FFFFFFF;
-
-  return l_diff;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n) {
-  int bits;
-
-  if (0xFFFF0000 & n) {
-    bits = 16;
-  } else {
-    bits = 0;
-  }
-  if (0x0000FF00 & (n >> bits)) bits += 8;
-  if (0x000000F0 & (n >> bits)) bits += 4;
-  if (0x0000000C & (n >> bits)) bits += 2;
-  if (0x00000002 & (n >> bits)) bits += 1;
-  if (0x00000001 & (n >> bits)) bits += 1;
-
-  return bits;
-}
-
-static __inline int WebRtcSpl_NormW32(WebRtc_Word32 a) {
-  int zeros;
-
-  if (a <= 0) a ^= 0xFFFFFFFF;
-
-  if (!(0xFFFF8000 & a)) {
-    zeros = 16;
-  } else {
-    zeros = 0;
-  }
-  if (!(0xFF800000 & (a << zeros))) zeros += 8;
-  if (!(0xF8000000 & (a << zeros))) zeros += 4;
-  if (!(0xE0000000 & (a << zeros))) zeros += 2;
-  if (!(0xC0000000 & (a << zeros))) zeros += 1;
-
-  return zeros;
-}
-
-static __inline int WebRtcSpl_NormU32(WebRtc_UWord32 a) {
-  int zeros;
-
-  if (a == 0) return 0;
-
-  if (!(0xFFFF0000 & a)) {
-    zeros = 16;
-  } else {
-    zeros = 0;
-  }
-  if (!(0xFF000000 & (a << zeros))) zeros += 8;
-  if (!(0xF0000000 & (a << zeros))) zeros += 4;
-  if (!(0xC0000000 & (a << zeros))) zeros += 2;
-  if (!(0x80000000 & (a << zeros))) zeros += 1;
-
-  return zeros;
-}
-
-static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
-  int zeros;
-
-  if (a <= 0) a ^= 0xFFFF;
-
-  if (!(0xFF80 & a)) {
-    zeros = 8;
-  } else {
-    zeros = 0;
-  }
-  if (!(0xF800 & (a << zeros))) zeros += 4;
-  if (!(0xE000 & (a << zeros))) zeros += 2;
-  if (!(0xC000 & (a << zeros))) zeros += 1;
-
-  return zeros;
-}
-
-static __inline int32_t WebRtc_MulAccumW16(int16_t a,
-                                          int16_t b,
-                                          int32_t c) {
-  return (a * b + c);
-}
-
-#endif  // WEBRTC_ARCH_ARM_V7A
-
-#endif  // WEBRTC_SPL_SPL_INL_H_

src/common_audio/signal_processing_library/main/interface/spl_inl_armv7.h

@@ -1,137 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-// This header file includes the inline functions for ARM processors in
-// the fix point signal processing library.
-
-#ifndef WEBRTC_SPL_SPL_INL_ARMV7_H_
-#define WEBRTC_SPL_SPL_INL_ARMV7_H_
-
-static __inline WebRtc_Word32 WEBRTC_SPL_MUL_16_32_RSFT16(WebRtc_Word16 a,
-                                                          WebRtc_Word32 b) {
-  WebRtc_Word32 tmp;
-  __asm__("smulwb %0, %1, %2":"=r"(tmp):"r"(b), "r"(a));
-  return tmp;
-}
-
-static __inline WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32(WebRtc_Word16 a,
-                                                          WebRtc_Word16 b,
-                                                          WebRtc_Word32 c) {
-  WebRtc_Word32 tmp;
-  __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(tmp) : "r"(b), "r"(a));
-  __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(tmp), "r"(c));
-  return tmp;
-}
-
-static __inline WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32BI(WebRtc_Word32 a,
-                                                            WebRtc_Word32 b) {
-  WebRtc_Word32 tmp;
-  __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
-  return tmp;
-}
-
-static __inline WebRtc_Word32 WEBRTC_SPL_MUL_16_16(WebRtc_Word16 a,
-                                                   WebRtc_Word16 b) {
-  WebRtc_Word32 tmp;
-  __asm__("smulbb %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
-  return tmp;
-}
-
-static __inline int32_t WebRtc_MulAccumW16(int16_t a,
-                                          int16_t b,
-                                          int32_t c) {
-  int32_t tmp = 0;
-  __asm__("smlabb %0, %1, %2, %3":"=r"(tmp):"r"(a), "r"(b), "r"(c));
-  return tmp;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
-                                                  WebRtc_Word16 b) {
-  WebRtc_Word32 s_sum;
-
-  __asm__("qadd16 %0, %1, %2":"=r"(s_sum):"r"(a), "r"(b));
-
-  return (WebRtc_Word16) s_sum;
-}
-
-static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
-                                                  WebRtc_Word32 l_var2) {
-  WebRtc_Word32 l_sum;
-
-  __asm__("qadd %0, %1, %2":"=r"(l_sum):"r"(l_var1), "r"(l_var2));
-
-  return l_sum;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
-                                                  WebRtc_Word16 var2) {
-  WebRtc_Word32 s_sub;
-
-  __asm__("qsub16 %0, %1, %2":"=r"(s_sub):"r"(var1), "r"(var2));
-
-  return (WebRtc_Word16)s_sub;
-}
-
-static __inline WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
-                                                  WebRtc_Word32 l_var2) {
-  WebRtc_Word32 l_sub;
-
-  __asm__("qsub %0, %1, %2":"=r"(l_sub):"r"(l_var1), "r"(l_var2));
-
-  return l_sub;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n) {
-  WebRtc_Word32 tmp;
-
-  __asm__("clz %0, %1":"=r"(tmp):"r"(n));
-
-  return (WebRtc_Word16)(32 - tmp);
-}
-
-static __inline int WebRtcSpl_NormW32(WebRtc_Word32 a) {
-  WebRtc_Word32 tmp;
-
-  if (a <= 0) a ^= 0xFFFFFFFF;
-
-  __asm__("clz %0, %1":"=r"(tmp):"r"(a));
-
-  return tmp - 1;
-}
-
-static __inline int WebRtcSpl_NormU32(WebRtc_UWord32 a) {
-  int tmp;
-
-  if (a == 0) return 0;
-
-  __asm__("clz %0, %1":"=r"(tmp):"r"(a));
-
-  return tmp;
-}
-
-static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
-  WebRtc_Word32 tmp;
-
-  if (a <= 0) a ^= 0xFFFFFFFF;
-
-  __asm__("clz %0, %1":"=r"(tmp):"r"(a));
-
-  return tmp - 17;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
-  WebRtc_Word16 out16;
-
-  __asm__("ssat %r0, #16, %r1" : "=r"(out16) : "r"(value32));
-
-  return out16;
-}
-#endif  // WEBRTC_SPL_SPL_INL_ARMV7_H_

src/common_audio/signal_processing_library/main/source/auto_correlation.c

@@ -1,141 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_AutoCorrelation().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-int WebRtcSpl_AutoCorrelation(G_CONST WebRtc_Word16* in_vector,
-                              int in_vector_length,
-                              int order,
-                              WebRtc_Word32* result,
-                              int* scale)
-{
-    WebRtc_Word32 sum;
-    int i, j;
-    WebRtc_Word16 smax; // Sample max
-    G_CONST WebRtc_Word16* xptr1;
-    G_CONST WebRtc_Word16* xptr2;
-    WebRtc_Word32* resultptr;
-    int scaling = 0;
-
-#ifdef _ARM_OPT_
-#pragma message("NOTE: _ARM_OPT_ optimizations are used")
-    WebRtc_Word16 loops4;
-#endif
-
-    if (order < 0)
-        order = in_vector_length;
-
-    // Find the max. sample
-    smax = WebRtcSpl_MaxAbsValueW16(in_vector, in_vector_length);
-
-    // In order to avoid overflow when computing the sum we should scale the samples so that
-    // (in_vector_length * smax * smax) will not overflow.
-
-    if (smax == 0)
-    {
-        scaling = 0;
-    } else
-    {
-        int nbits = WebRtcSpl_GetSizeInBits(in_vector_length); // # of bits in the sum loop
-        int t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax)); // # of bits to normalize smax
-
-        if (t > nbits)
-        {
-            scaling = 0;
-        } else
-        {
-            scaling = nbits - t;
-        }
-
-    }
-
-    resultptr = result;
-
-    // Perform the actual correlation calculation
-    for (i = 0; i < order + 1; i++)
-    {
-        int loops = (in_vector_length - i);
-        sum = 0;
-        xptr1 = in_vector;
-        xptr2 = &in_vector[i];
-#ifndef _ARM_OPT_
-        for (j = loops; j > 0; j--)
-        {
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1++, *xptr2++, scaling);
-        }
-#else
-        loops4 = (loops >> 2) << 2;
-
-        if (scaling == 0)
-        {
-            for (j = 0; j < loops4; j = j + 4)
-            {
-                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
-                xptr1++;
-                xptr2++;
-            }
-
-            for (j = loops4; j < loops; j++)
-            {
-                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
-                xptr1++;
-                xptr2++;
-            }
-        }
-        else
-        {
-            for (j = 0; j < loops4; j = j + 4)
-            {
-                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
-                xptr1++;
-                xptr2++;
-            }
-
-            for (j = loops4; j < loops; j++)
-            {
-                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
-                xptr1++;
-                xptr2++;
-            }
-        }
-
-#endif
-        *resultptr++ = sum;
-    }
-
-    *scale = scaling;
-
-    return order + 1;
-}

src/common_audio/signal_processing_library/main/source/complex_bit_reverse.c

@@ -1,51 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_ComplexBitReverse().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_ComplexBitReverse(WebRtc_Word16 frfi[], int stages)
-{
-    int mr, nn, n, l, m;
-    WebRtc_Word16 tr, ti;
-
-    n = 1 << stages;
-
-    mr = 0;
-    nn = n - 1;
-
-    // decimation in time - re-order data
-    for (m = 1; m <= nn; ++m)
-    {
-        l = n;
-        do
-        {
-            l >>= 1;
-        } while (mr + l > nn);
-        mr = (mr & (l - 1)) + l;
-
-        if (mr <= m)
-            continue;
-
-        tr = frfi[2 * m];
-        frfi[2 * m] = frfi[2 * mr];
-        frfi[2 * mr] = tr;
-
-        ti = frfi[2 * m + 1];
-        frfi[2 * m + 1] = frfi[2 * mr + 1];
-        frfi[2 * mr + 1] = ti;
-    }
-}

src/common_audio/signal_processing_library/main/source/complex_fft.c

@@ -1,150 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_ComplexFFT().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define CFFTSFT 14
-#define CFFTRND 1
-#define CFFTRND2 16384
-
-int WebRtcSpl_ComplexFFT(WebRtc_Word16 frfi[], int stages, int mode)
-{
-    int i, j, l, k, istep, n, m;
-    WebRtc_Word16 wr, wi;
-    WebRtc_Word32 tr32, ti32, qr32, qi32;
-
-    /* The 1024-value is a constant given from the size of WebRtcSpl_kSinTable1024[],
-     * and should not be changed depending on the input parameter 'stages'
-     */
-    n = 1 << stages;
-    if (n > 1024)
-        return -1;
-
-    l = 1;
-    k = 10 - 1; /* Constant for given WebRtcSpl_kSinTable1024[]. Do not change
-         depending on the input parameter 'stages' */
-
-    if (mode == 0)
-    {
-        // mode==0: Low-complexity and Low-accuracy mode
-        while (l < n)
-        {
-            istep = l << 1;
-
-            for (m = 0; m < l; ++m)
-            {
-                j = m << k;
-
-                /* The 256-value is a constant given as 1/4 of the size of
-                 * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
-                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
-                 */
-                wr = WebRtcSpl_kSinTable1024[j + 256];
-                wi = -WebRtcSpl_kSinTable1024[j];
-
-                for (i = m; i < n; i += istep)
-                {
-                    j = i + l;
-
-                    tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
-                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1])), 15);
-
-                    ti32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
-                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j])), 15);
-
-                    qr32 = (WebRtc_Word32)frfi[2 * i];
-                    qi32 = (WebRtc_Word32)frfi[2 * i + 1];
-                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, 1);
-                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, 1);
-                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, 1);
-                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, 1);
-                }
-            }
-
-            --k;
-            l = istep;
-
-        }
-
-    } else
-    {
-        // mode==1: High-complexity and High-accuracy mode
-        while (l < n)
-        {
-            istep = l << 1;
-
-            for (m = 0; m < l; ++m)
-            {
-                j = m << k;
-
-                /* The 256-value is a constant given as 1/4 of the size of
-                 * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
-                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
-                 */
-                wr = WebRtcSpl_kSinTable1024[j + 256];
-                wi = -WebRtcSpl_kSinTable1024[j];
-
-#ifdef WEBRTC_ARCH_ARM_V7A
-                WebRtc_Word32 wri;
-                WebRtc_Word32 frfi_r;
-                __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
-                    "r"((WebRtc_Word32)wr), "r"((WebRtc_Word32)wi));
-#endif
-
-                for (i = m; i < n; i += istep)
-                {
-                    j = i + l;
-
-#ifdef WEBRTC_ARCH_ARM_V7A
-                    __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(frfi_r) :
-                        "r"((WebRtc_Word32)frfi[2*j]), "r"((WebRtc_Word32)frfi[2*j +1]));
-                    __asm__("smlsd %0, %1, %2, %3" : "=r"(tr32) :
-                        "r"(wri), "r"(frfi_r), "r"(CFFTRND));
-                    __asm__("smladx %0, %1, %2, %3" : "=r"(ti32) :
-                        "r"(wri), "r"(frfi_r), "r"(CFFTRND));
-    
-#else
-                    tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
-                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CFFTRND;
-
-                    ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
-                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CFFTRND;
-#endif
-
-                    tr32 = WEBRTC_SPL_RSHIFT_W32(tr32, 15 - CFFTSFT);
-                    ti32 = WEBRTC_SPL_RSHIFT_W32(ti32, 15 - CFFTSFT);
-
-                    qr32 = ((WebRtc_Word32)frfi[2 * i]) << CFFTSFT;
-                    qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CFFTSFT;
-
-                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qr32 - tr32 + CFFTRND2), 1 + CFFTSFT);
-                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qi32 - ti32 + CFFTRND2), 1 + CFFTSFT);
-                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qr32 + tr32 + CFFTRND2), 1 + CFFTSFT);
-                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qi32 + ti32 + CFFTRND2), 1 + CFFTSFT);
-                }
-            }
-
-            --k;
-            l = istep;
-        }
-    }
-    return 0;
-}

src/common_audio/signal_processing_library/main/source/complex_ifft.c

@@ -1,161 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_ComplexIFFT().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define CIFFTSFT 14
-#define CIFFTRND 1
-
-int WebRtcSpl_ComplexIFFT(WebRtc_Word16 frfi[], int stages, int mode)
-{
-    int i, j, l, k, istep, n, m, scale, shift;
-    WebRtc_Word16 wr, wi;
-    WebRtc_Word32 tr32, ti32, qr32, qi32;
-    WebRtc_Word32 tmp32, round2;
-
-    /* The 1024-value is a constant given from the size of WebRtcSpl_kSinTable1024[],
-     * and should not be changed depending on the input parameter 'stages'
-     */
-    n = 1 << stages;
-    if (n > 1024)
-        return -1;
-
-    scale = 0;
-
-    l = 1;
-    k = 10 - 1; /* Constant for given WebRtcSpl_kSinTable1024[]. Do not change
-         depending on the input parameter 'stages' */
-
-    while (l < n)
-    {
-        // variable scaling, depending upon data
-        shift = 0;
-        round2 = 8192;
-
-        tmp32 = (WebRtc_Word32)WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
-        if (tmp32 > 13573)
-        {
-            shift++;
-            scale++;
-            round2 <<= 1;
-        }
-        if (tmp32 > 27146)
-        {
-            shift++;
-            scale++;
-            round2 <<= 1;
-        }
-
-        istep = l << 1;
-
-        if (mode == 0)
-        {
-            // mode==0: Low-complexity and Low-accuracy mode
-            for (m = 0; m < l; ++m)
-            {
-                j = m << k;
-
-                /* The 256-value is a constant given as 1/4 of the size of
-                 * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
-                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
-                 */
-                wr = WebRtcSpl_kSinTable1024[j + 256];
-                wi = WebRtcSpl_kSinTable1024[j];
-
-                for (i = m; i < n; i += istep)
-                {
-                    j = i + l;
-
-                    tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j], 0)
-                            - WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j + 1], 0)), 15);
-
-                    ti32 = WEBRTC_SPL_RSHIFT_W32(
-                            (WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j + 1], 0)
-                                    + WEBRTC_SPL_MUL_16_16_RSFT(wi,frfi[2*j],0)), 15);
-
-                    qr32 = (WebRtc_Word32)frfi[2 * i];
-                    qi32 = (WebRtc_Word32)frfi[2 * i + 1];
-                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, shift);
-                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, shift);
-                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, shift);
-                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, shift);
-                }
-            }
-        } else
-        {
-            // mode==1: High-complexity and High-accuracy mode
-
-            for (m = 0; m < l; ++m)
-            {
-                j = m << k;
-
-                /* The 256-value is a constant given as 1/4 of the size of
-                 * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
-                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
-                 */
-                wr = WebRtcSpl_kSinTable1024[j + 256];
-                wi = WebRtcSpl_kSinTable1024[j];
-
-#ifdef WEBRTC_ARCH_ARM_V7A
-                WebRtc_Word32 wri;
-                WebRtc_Word32 frfi_r;
-                __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
-                    "r"((WebRtc_Word32)wr), "r"((WebRtc_Word32)wi));
-#endif
-
-                for (i = m; i < n; i += istep)
-                {
-                    j = i + l;
-
-#ifdef WEBRTC_ARCH_ARM_V7A
-                    __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(frfi_r) :
-                        "r"((WebRtc_Word32)frfi[2*j]), "r"((WebRtc_Word32)frfi[2*j +1]));
-                    __asm__("smlsd %0, %1, %2, %3" : "=r"(tr32) :
-                        "r"(wri), "r"(frfi_r), "r"(CIFFTRND));
-                    __asm__("smladx %0, %1, %2, %3" : "=r"(ti32) :
-                        "r"(wri), "r"(frfi_r), "r"(CIFFTRND));
-#else
-
-                    tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
-                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CIFFTRND;
-
-                    ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
-                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CIFFTRND;
-#endif
-                    tr32 = WEBRTC_SPL_RSHIFT_W32(tr32, 15 - CIFFTSFT);
-                    ti32 = WEBRTC_SPL_RSHIFT_W32(ti32, 15 - CIFFTSFT);
-    
-                    qr32 = ((WebRtc_Word32)frfi[2 * i]) << CIFFTSFT;
-                    qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CIFFTSFT;
-    
-                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 - tr32+round2),
-                                                                       shift+CIFFTSFT);
-                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qi32 - ti32 + round2), shift + CIFFTSFT);
-                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 + tr32 + round2),
-                                                                       shift + CIFFTSFT);
-                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qi32 + ti32 + round2), shift + CIFFTSFT);
-                }
-            }
-
-        }
-        --k;
-        l = istep;
-    }
-    return scale;
-}

src/common_audio/signal_processing_library/main/source/copy_set_operations.c

@@ -1,108 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the implementation of functions
- * WebRtcSpl_MemSetW16()
- * WebRtcSpl_MemSetW32()
- * WebRtcSpl_MemCpyReversedOrder()
- * WebRtcSpl_CopyFromEndW16()
- * WebRtcSpl_ZerosArrayW16()
- * WebRtcSpl_ZerosArrayW32()
- * WebRtcSpl_OnesArrayW16()
- * WebRtcSpl_OnesArrayW32()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include <string.h>
-#include "signal_processing_library.h"
-
-
-void WebRtcSpl_MemSetW16(WebRtc_Word16 *ptr, WebRtc_Word16 set_value, int length)
-{
-    int j;
-    WebRtc_Word16 *arrptr = ptr;
-
-    for (j = length; j > 0; j--)
-    {
-        *arrptr++ = set_value;
-    }
-}
-
-void WebRtcSpl_MemSetW32(WebRtc_Word32 *ptr, WebRtc_Word32 set_value, int length)
-{
-    int j;
-    WebRtc_Word32 *arrptr = ptr;
-
-    for (j = length; j > 0; j--)
-    {
-        *arrptr++ = set_value;
-    }
-}
-
-void WebRtcSpl_MemCpyReversedOrder(WebRtc_Word16* dest, WebRtc_Word16* source, int length)
-{
-    int j;
-    WebRtc_Word16* destPtr = dest;
-    WebRtc_Word16* sourcePtr = source;
-
-    for (j = 0; j < length; j++)
-    {
-        *destPtr-- = *sourcePtr++;
-    }
-}
-
-WebRtc_Word16 WebRtcSpl_CopyFromEndW16(G_CONST WebRtc_Word16 *vector_in,
-                                       WebRtc_Word16 length,
-                                       WebRtc_Word16 samples,
-                                       WebRtc_Word16 *vector_out)
-{
-    // Copy the last <samples> of the input vector to vector_out
-    WEBRTC_SPL_MEMCPY_W16(vector_out, &vector_in[length - samples], samples);
-
-    return samples;
-}
-
-WebRtc_Word16 WebRtcSpl_ZerosArrayW16(WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
-    WebRtcSpl_MemSetW16(vector, 0, length);
-    return length;
-}
-
-WebRtc_Word16 WebRtcSpl_ZerosArrayW32(WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
-    WebRtcSpl_MemSetW32(vector, 0, length);
-    return length;
-}
-
-WebRtc_Word16 WebRtcSpl_OnesArrayW16(WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 i;
-    WebRtc_Word16 *tmpvec = vector;
-    for (i = 0; i < length; i++)
-    {
-        *tmpvec++ = 1;
-    }
-    return length;
-}
-
-WebRtc_Word16 WebRtcSpl_OnesArrayW32(WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 i;
-    WebRtc_Word32 *tmpvec = vector;
-    for (i = 0; i < length; i++)
-    {
-        *tmpvec++ = 1;
-    }
-    return length;
-}

src/common_audio/signal_processing_library/main/source/cos_table.c

@@ -1,60 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the 360 degree cos table.
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_kCosTable[] = {
-        8192,  8190,  8187,  8180,  8172,  8160,  8147,  8130,  8112,
-        8091,  8067,  8041,  8012,  7982,  7948,  7912,  7874,  7834,
-        7791,  7745,  7697,  7647,  7595,  7540,  7483,  7424,  7362,
-        7299,  7233,  7164,  7094,  7021,  6947,  6870,  6791,  6710,
-        6627,  6542,  6455,  6366,  6275,  6182,  6087,  5991,  5892,
-        5792,  5690,  5586,  5481,  5374,  5265,  5155,  5043,  4930,
-        4815,  4698,  4580,  4461,  4341,  4219,  4096,  3971,  3845,
-        3719,  3591,  3462,  3331,  3200,  3068,  2935,  2801,  2667,
-        2531,  2395,  2258,  2120,  1981,  1842,  1703,  1563,  1422,
-        1281,  1140,   998,   856,   713,   571,   428,   285,   142,
-           0,  -142,  -285,  -428,  -571,  -713,  -856,  -998, -1140,
-       -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395,
-       -2531, -2667, -2801, -2935, -3068, -3200, -3331, -3462, -3591,
-       -3719, -3845, -3971, -4095, -4219, -4341, -4461, -4580, -4698,
-       -4815, -4930, -5043, -5155, -5265, -5374, -5481, -5586, -5690,
-       -5792, -5892, -5991, -6087, -6182, -6275, -6366, -6455, -6542,
-       -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164, -7233,
-       -7299, -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745,
-       -7791, -7834, -7874, -7912, -7948, -7982, -8012, -8041, -8067,
-       -8091, -8112, -8130, -8147, -8160, -8172, -8180, -8187, -8190,
-       -8191, -8190, -8187, -8180, -8172, -8160, -8147, -8130, -8112,
-       -8091, -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
-       -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362,
-       -7299, -7233, -7164, -7094, -7021, -6947, -6870, -6791, -6710,
-       -6627, -6542, -6455, -6366, -6275, -6182, -6087, -5991, -5892,
-       -5792, -5690, -5586, -5481, -5374, -5265, -5155, -5043, -4930,
-       -4815, -4698, -4580, -4461, -4341, -4219, -4096, -3971, -3845,
-       -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801, -2667,
-       -2531, -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422,
-       -1281, -1140,  -998,  -856,  -713,  -571,  -428,  -285,  -142,
-           0,   142,   285,   428,   571,   713,   856,   998,  1140,
-        1281,  1422,  1563,  1703,  1842,  1981,  2120,  2258,  2395,
-        2531,  2667,  2801,  2935,  3068,  3200,  3331,  3462,  3591,
-        3719,  3845,  3971,  4095,  4219,  4341,  4461,  4580,  4698,
-        4815,  4930,  5043,  5155,  5265,  5374,  5481,  5586,  5690,
-        5792,  5892,  5991,  6087,  6182,  6275,  6366,  6455,  6542,
-        6627,  6710,  6791,  6870,  6947,  7021,  7094,  7164,  7233,
-        7299,  7362,  7424,  7483,  7540,  7595,  7647,  7697,  7745,
-        7791,  7834,  7874,  7912,  7948,  7982,  8012,  8041,  8067,
-        8091,  8112,  8130,  8147,  8160,  8172,  8180,  8187,  8190
-};

src/common_audio/signal_processing_library/main/source/cross_correlation.c

@@ -1,267 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_CrossCorrelation().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_correlation, WebRtc_Word16* seq1,
-                                WebRtc_Word16* seq2, WebRtc_Word16 dim_seq,
-                                WebRtc_Word16 dim_cross_correlation,
-                                WebRtc_Word16 right_shifts,
-                                WebRtc_Word16 step_seq2)
-{
-    int i, j;
-    WebRtc_Word16* seq1Ptr;
-    WebRtc_Word16* seq2Ptr;
-    WebRtc_Word32* CrossCorrPtr;
-
-#ifdef _XSCALE_OPT_
-
-#ifdef _WIN32
-#pragma message("NOTE: _XSCALE_OPT_ optimizations are used (overrides _ARM_OPT_ and requires /QRxscale compiler flag)")
-#endif
-
-    __int64 macc40;
-
-    int iseq1[250];
-    int iseq2[250];
-    int iseq3[250];
-    int * iseq1Ptr;
-    int * iseq2Ptr;
-    int * iseq3Ptr;
-    int len, i_len;
-
-    seq1Ptr = seq1;
-    iseq1Ptr = iseq1;
-    for(i = 0; i < ((dim_seq + 1) >> 1); i++)
-    {
-        *iseq1Ptr = (unsigned short)*seq1Ptr++;
-        *iseq1Ptr++ |= (WebRtc_Word32)*seq1Ptr++ << 16;
-
-    }
-
-    if(dim_seq%2)
-    {
-        *(iseq1Ptr-1) &= 0x0000ffff;
-    }
-    *iseq1Ptr = 0;
-    iseq1Ptr++;
-    *iseq1Ptr = 0;
-    iseq1Ptr++;
-    *iseq1Ptr = 0;
-
-    if(step_seq2 < 0)
-    {
-        seq2Ptr = seq2 - dim_cross_correlation + 1;
-        CrossCorrPtr = &cross_correlation[dim_cross_correlation - 1];
-    }
-    else
-    {
-        seq2Ptr = seq2;
-        CrossCorrPtr = cross_correlation;
-    }
-
-    len = dim_seq + dim_cross_correlation - 1;
-    i_len = (len + 1) >> 1;
-    iseq2Ptr = iseq2;
-
-    iseq3Ptr = iseq3;
-    for(i = 0; i < i_len; i++)
-    {
-        *iseq2Ptr = (unsigned short)*seq2Ptr++;
-        *iseq3Ptr = (unsigned short)*seq2Ptr;
-        *iseq2Ptr++ |= (WebRtc_Word32)*seq2Ptr++ << 16;
-        *iseq3Ptr++ |= (WebRtc_Word32)*seq2Ptr << 16;
-    }
-
-    if(len % 2)
-    {
-        iseq2[i_len - 1] &= 0x0000ffff;
-        iseq3[i_len - 1] = 0;
-    }
-    else
-    iseq3[i_len - 1] &= 0x0000ffff;
-
-    iseq2[i_len] = 0;
-    iseq3[i_len] = 0;
-    iseq2[i_len + 1] = 0;
-    iseq3[i_len + 1] = 0;
-    iseq2[i_len + 2] = 0;
-    iseq3[i_len + 2] = 0;
-
-    // Set pointer to start value
-    iseq2Ptr = iseq2;
-    iseq3Ptr = iseq3;
-
-    i_len = (dim_seq + 7) >> 3;
-    for (i = 0; i < dim_cross_correlation; i++)
-    {
-
-        iseq1Ptr = iseq1;
-
-        macc40 = 0;
-
-        _WriteCoProcessor(macc40, 0);
-
-        if((i & 1))
-        {
-            iseq3Ptr = iseq3 + (i >> 1);
-            for (j = i_len; j > 0; j--)
-            {
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
-            }
-        }
-        else
-        {
-            iseq2Ptr = iseq2 + (i >> 1);
-            for (j = i_len; j > 0; j--)
-            {
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
-            }
-
-        }
-
-        macc40 = _ReadCoProcessor(0);
-        *CrossCorrPtr = (WebRtc_Word32)(macc40 >> right_shifts);
-        CrossCorrPtr += step_seq2;
-    }
-#else // #ifdef _XSCALE_OPT_
-#ifdef _ARM_OPT_
-    WebRtc_Word16 dim_seq8 = (dim_seq >> 3) << 3;
-#endif
-
-    CrossCorrPtr = cross_correlation;
-
-    for (i = 0; i < dim_cross_correlation; i++)
-    {
-        // Set the pointer to the static vector, set the pointer to the sliding vector
-        // and initialize cross_correlation
-        seq1Ptr = seq1;
-        seq2Ptr = seq2 + (step_seq2 * i);
-        (*CrossCorrPtr) = 0;
-
-#ifndef _ARM_OPT_ 
-#ifdef _WIN32
-#pragma message("NOTE: default implementation is used")
-#endif
-        // Perform the cross correlation
-        for (j = 0; j < dim_seq; j++)
-        {
-            (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr), right_shifts);
-            seq1Ptr++;
-            seq2Ptr++;
-        }
-#else
-#ifdef _WIN32
-#pragma message("NOTE: _ARM_OPT_ optimizations are used")
-#endif
-        if (right_shifts == 0)
-        {
-            // Perform the optimized cross correlation
-            for (j = 0; j < dim_seq8; j = j + 8)
-            {
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-            }
-
-            for (j = dim_seq8; j < dim_seq; j++)
-            {
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-            }
-        }
-        else // right_shifts != 0
-
-        {
-            // Perform the optimized cross correlation
-            for (j = 0; j < dim_seq8; j = j + 8)
-            {
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-            }
-
-            for (j = dim_seq8; j < dim_seq; j++)
-            {
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-            }
-        }
-#endif
-        CrossCorrPtr++;
-    }
-#endif
-}

src/common_audio/signal_processing_library/main/source/division_operations.c

@@ -1,144 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the divisions
- * WebRtcSpl_DivU32U16()
- * WebRtcSpl_DivW32W16()
- * WebRtcSpl_DivW32W16ResW16()
- * WebRtcSpl_DivResultInQ31()
- * WebRtcSpl_DivW32HiLow()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_UWord32 WebRtcSpl_DivU32U16(WebRtc_UWord32 num, WebRtc_UWord16 den)
-{
-    // Guard against division with 0
-    if (den != 0)
-    {
-        return (WebRtc_UWord32)(num / den);
-    } else
-    {
-        return (WebRtc_UWord32)0xFFFFFFFF;
-    }
-}
-
-WebRtc_Word32 WebRtcSpl_DivW32W16(WebRtc_Word32 num, WebRtc_Word16 den)
-{
-    // Guard against division with 0
-    if (den != 0)
-    {
-        return (WebRtc_Word32)(num / den);
-    } else
-    {
-        return (WebRtc_Word32)0x7FFFFFFF;
-    }
-}
-
-WebRtc_Word16 WebRtcSpl_DivW32W16ResW16(WebRtc_Word32 num, WebRtc_Word16 den)
-{
-    // Guard against division with 0
-    if (den != 0)
-    {
-        return (WebRtc_Word16)(num / den);
-    } else
-    {
-        return (WebRtc_Word16)0x7FFF;
-    }
-}
-
-WebRtc_Word32 WebRtcSpl_DivResultInQ31(WebRtc_Word32 num, WebRtc_Word32 den)
-{
-    WebRtc_Word32 L_num = num;
-    WebRtc_Word32 L_den = den;
-    WebRtc_Word32 div = 0;
-    int k = 31;
-    int change_sign = 0;
-
-    if (num == 0)
-        return 0;
-
-    if (num < 0)
-    {
-        change_sign++;
-        L_num = -num;
-    }
-    if (den < 0)
-    {
-        change_sign++;
-        L_den = -den;
-    }
-    while (k--)
-    {
-        div <<= 1;
-        L_num <<= 1;
-        if (L_num >= L_den)
-        {
-            L_num -= L_den;
-            div++;
-        }
-    }
-    if (change_sign == 1)
-    {
-        div = -div;
-    }
-    return div;
-}
-
-WebRtc_Word32 WebRtcSpl_DivW32HiLow(WebRtc_Word32 num, WebRtc_Word16 den_hi,
-                                    WebRtc_Word16 den_low)
-{
-    WebRtc_Word16 approx, tmp_hi, tmp_low, num_hi, num_low;
-    WebRtc_Word32 tmpW32;
-
-    approx = (WebRtc_Word16)WebRtcSpl_DivW32W16((WebRtc_Word32)0x1FFFFFFF, den_hi);
-    // result in Q14 (Note: 3FFFFFFF = 0.5 in Q30)
-
-    // tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30)
-    tmpW32 = (WEBRTC_SPL_MUL_16_16(den_hi, approx) << 1)
-            + ((WEBRTC_SPL_MUL_16_16(den_low, approx) >> 15) << 1);
-    // tmpW32 = den * approx
-
-    tmpW32 = (WebRtc_Word32)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx))
-
-    // Store tmpW32 in hi and low format
-    tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
-    tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
-    // tmpW32 = 1/den in Q29
-    tmpW32 = ((WEBRTC_SPL_MUL_16_16(tmp_hi, approx) + (WEBRTC_SPL_MUL_16_16(tmp_low, approx)
-            >> 15)) << 1);
-
-    // 1/den in hi and low format
-    tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
-    tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
-    // Store num in hi and low format
-    num_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(num, 16);
-    num_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((num
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)num_hi, 16)), 1);
-
-    // num * (1/den) by 32 bit multiplication (result in Q28)
-
-    tmpW32 = (WEBRTC_SPL_MUL_16_16(num_hi, tmp_hi) + (WEBRTC_SPL_MUL_16_16(num_hi, tmp_low)
-            >> 15) + (WEBRTC_SPL_MUL_16_16(num_low, tmp_hi) >> 15));
-
-    // Put result in Q31 (convert from Q28)
-    tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
-
-    return tmpW32;
-}

src/common_audio/signal_processing_library/main/source/dot_product_with_scale.c

@@ -1,91 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_DotProductWithScale().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word32 WebRtcSpl_DotProductWithScale(WebRtc_Word16 *vector1, WebRtc_Word16 *vector2,
-                                            int length, int scaling)
-{
-    WebRtc_Word32 sum;
-    int i;
-#ifdef _ARM_OPT_
-#pragma message("NOTE: _ARM_OPT_ optimizations are used")
-    WebRtc_Word16 len4 = (length >> 2) << 2;
-#endif
-
-    sum = 0;
-
-#ifndef _ARM_OPT_
-    for (i = 0; i < length; i++)
-    {
-        sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1++, *vector2++, scaling);
-    }
-#else
-    if (scaling == 0)
-    {
-        for (i = 0; i < len4; i = i + 4)
-        {
-            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
-            vector1++;
-            vector2++;
-        }
-
-        for (i = len4; i < length; i++)
-        {
-            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
-            vector1++;
-            vector2++;
-        }
-    }
-    else
-    {
-        for (i = 0; i < len4; i = i + 4)
-        {
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
-            vector1++;
-            vector2++;
-        }
-
-        for (i = len4; i < length; i++)
-        {
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
-            vector1++;
-            vector2++;
-        }
-    }
-#endif
-
-    return sum;
-}

src/common_audio/signal_processing_library/main/source/downsample_fast.c

@@ -1,59 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_DownsampleFast().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-int WebRtcSpl_DownsampleFast(WebRtc_Word16 *in_ptr, WebRtc_Word16 in_length,
-                             WebRtc_Word16 *out_ptr, WebRtc_Word16 out_length,
-                             WebRtc_Word16 *B, WebRtc_Word16 B_length, WebRtc_Word16 factor,
-                             WebRtc_Word16 delay)
-{
-    WebRtc_Word32 o;
-    int i, j;
-
-    WebRtc_Word16 *downsampled_ptr = out_ptr;
-    WebRtc_Word16 *b_ptr;
-    WebRtc_Word16 *x_ptr;
-    WebRtc_Word16 endpos = delay
-            + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(factor, (out_length - 1)) + 1;
-
-    if (in_length < endpos)
-    {
-        return -1;
-    }
-
-    for (i = delay; i < endpos; i += factor)
-    {
-        b_ptr = &B[0];
-        x_ptr = &in_ptr[i];
-
-        o = (WebRtc_Word32)2048; // Round val
-
-        for (j = 0; j < B_length; j++)
-        {
-            o += WEBRTC_SPL_MUL_16_16(*b_ptr++, *x_ptr--);
-        }
-
-        o = WEBRTC_SPL_RSHIFT_W32(o, 12);
-
-        // If output is higher than 32768, saturate it. Same with negative side
-
-        *downsampled_ptr++ = WebRtcSpl_SatW32ToW16(o);
-    }
-
-    return 0;
-}

src/common_audio/signal_processing_library/main/source/filter_ar.c

@@ -1,89 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_FilterAR().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* a,
-                       int a_length,
-                       G_CONST WebRtc_Word16* x,
-                       int x_length,
-                       WebRtc_Word16* state,
-                       int state_length,
-                       WebRtc_Word16* state_low,
-                       int state_low_length,
-                       WebRtc_Word16* filtered,
-                       WebRtc_Word16* filtered_low,
-                       int filtered_low_length)
-{
-    WebRtc_Word32 o;
-    WebRtc_Word32 oLOW;
-    int i, j, stop;
-    G_CONST WebRtc_Word16* x_ptr = &x[0];
-    WebRtc_Word16* filteredFINAL_ptr = filtered;
-    WebRtc_Word16* filteredFINAL_LOW_ptr = filtered_low;
-
-    for (i = 0; i < x_length; i++)
-    {
-        // Calculate filtered[i] and filtered_low[i]
-        G_CONST WebRtc_Word16* a_ptr = &a[1];
-        WebRtc_Word16* filtered_ptr = &filtered[i - 1];
-        WebRtc_Word16* filtered_low_ptr = &filtered_low[i - 1];
-        WebRtc_Word16* state_ptr = &state[state_length - 1];
-        WebRtc_Word16* state_low_ptr = &state_low[state_length - 1];
-
-        o = (WebRtc_Word32)(*x_ptr++) << 12;
-        oLOW = (WebRtc_Word32)0;
-
-        stop = (i < a_length) ? i + 1 : a_length;
-        for (j = 1; j < stop; j++)
-        {
-            o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *filtered_ptr--);
-            oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *filtered_low_ptr--);
-        }
-        for (j = i + 1; j < a_length; j++)
-        {
-            o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *state_ptr--);
-            oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *state_low_ptr--);
-        }
-
-        o += (oLOW >> 12);
-        *filteredFINAL_ptr = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
-        *filteredFINAL_LOW_ptr++ = (WebRtc_Word16)(o - ((WebRtc_Word32)(*filteredFINAL_ptr++)
-                << 12));
-    }
-
-    // Save the filter state
-    if (x_length >= state_length)
-    {
-        WebRtcSpl_CopyFromEndW16(filtered, x_length, a_length - 1, state);
-        WebRtcSpl_CopyFromEndW16(filtered_low, x_length, a_length - 1, state_low);
-    } else
-    {
-        for (i = 0; i < state_length - x_length; i++)
-        {
-            state[i] = state[i + x_length];
-            state_low[i] = state_low[i + x_length];
-        }
-        for (i = 0; i < x_length; i++)
-        {
-            state[state_length - x_length + i] = filtered[i];
-            state[state_length - x_length + i] = filtered_low[i];
-        }
-    }
-
-    return x_length;
-}

src/common_audio/signal_processing_library/main/source/filter_ar_fast_q12.c

@@ -1,49 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_FilterARFastQ12().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_FilterARFastQ12(WebRtc_Word16 *in, WebRtc_Word16 *out, WebRtc_Word16 *A,
-                               WebRtc_Word16 A_length, WebRtc_Word16 length)
-{
-    WebRtc_Word32 o;
-    int i, j;
-
-    WebRtc_Word16 *x_ptr = &in[0];
-    WebRtc_Word16 *filtered_ptr = &out[0];
-
-    for (i = 0; i < length; i++)
-    {
-        // Calculate filtered[i]
-        G_CONST WebRtc_Word16 *a_ptr = &A[0];
-        WebRtc_Word16 *state_ptr = &out[i - 1];
-
-        o = WEBRTC_SPL_MUL_16_16(*x_ptr++, *a_ptr++);
-
-        for (j = 1; j < A_length; j++)
-        {
-            o -= WEBRTC_SPL_MUL_16_16(*a_ptr++,*state_ptr--);
-        }
-
-        // Saturate the output
-        o = WEBRTC_SPL_SAT((WebRtc_Word32)134215679, o, (WebRtc_Word32)-134217728);
-
-        *filtered_ptr++ = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
-    }
-
-    return;
-}

src/common_audio/signal_processing_library/main/source/filter_ma_fast_q12.c

@@ -1,49 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_FilterMAFastQ12().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_FilterMAFastQ12(WebRtc_Word16* in_ptr,
-                               WebRtc_Word16* out_ptr,
-                               WebRtc_Word16* B,
-                               WebRtc_Word16 B_length,
-                               WebRtc_Word16 length)
-{
-    WebRtc_Word32 o;
-    int i, j;
-    for (i = 0; i < length; i++)
-    {
-        G_CONST WebRtc_Word16* b_ptr = &B[0];
-        G_CONST WebRtc_Word16* x_ptr = &in_ptr[i];
-
-        o = (WebRtc_Word32)0;
-
-        for (j = 0; j < B_length; j++)
-        {
-            o += WEBRTC_SPL_MUL_16_16(*b_ptr++, *x_ptr--);
-        }
-
-        // If output is higher than 32768, saturate it. Same with negative side
-        // 2^27 = 134217728, which corresponds to 32768 in Q12
-
-        // Saturate the output
-        o = WEBRTC_SPL_SAT((WebRtc_Word32)134215679, o, (WebRtc_Word32)-134217728);
-
-        *out_ptr++ = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
-    }
-    return;
-}

src/common_audio/signal_processing_library/main/source/get_hanning_window.c

@@ -1,41 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_GetHanningWindow().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_GetHanningWindow(WebRtc_Word16 *v, WebRtc_Word16 size)
-{
-    int jj;
-    WebRtc_Word16 *vptr1;
-
-    WebRtc_Word32 index;
-    WebRtc_Word32 factor = ((WebRtc_Word32)0x40000000);
-
-    factor = WebRtcSpl_DivW32W16(factor, size);
-    if (size < 513)
-        index = (WebRtc_Word32)-0x200000;
-    else
-        index = (WebRtc_Word32)-0x100000;
-    vptr1 = v;
-
-    for (jj = 0; jj < size; jj++)
-    {
-        index += factor;
-        (*vptr1++) = WebRtcSpl_kHanningTable[index >> 22];
-    }
-
-}

src/common_audio/signal_processing_library/main/source/ilbc_specific_functions.c

@@ -1,120 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the iLBC specific functions
- * WebRtcSpl_ScaleAndAddVectorsWithRound()
- * WebRtcSpl_ReverseOrderMultArrayElements()
- * WebRtcSpl_ElementwiseVectorMult()
- * WebRtcSpl_AddVectorsAndShift()
- * WebRtcSpl_AddAffineVectorToVector()
- * WebRtcSpl_AffineTransformVector()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_ScaleAndAddVectorsWithRound(WebRtc_Word16 *vector1, WebRtc_Word16 scale1,
-                                           WebRtc_Word16 *vector2, WebRtc_Word16 scale2,
-                                           WebRtc_Word16 right_shifts, WebRtc_Word16 *out,
-                                           WebRtc_Word16 vector_length)
-{
-    int i;
-    WebRtc_Word16 roundVal;
-    roundVal = 1 << right_shifts;
-    roundVal = roundVal >> 1;
-    for (i = 0; i < vector_length; i++)
-    {
-        out[i] = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(vector1[i], scale1)
-                + WEBRTC_SPL_MUL_16_16(vector2[i], scale2) + roundVal) >> right_shifts);
-    }
-}
-
-void WebRtcSpl_ReverseOrderMultArrayElements(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in,
-                                             G_CONST WebRtc_Word16 *win,
-                                             WebRtc_Word16 vector_length,
-                                             WebRtc_Word16 right_shifts)
-{
-    int i;
-    WebRtc_Word16 *outptr = out;
-    G_CONST WebRtc_Word16 *inptr = in;
-    G_CONST WebRtc_Word16 *winptr = win;
-    for (i = 0; i < vector_length; i++)
-    {
-        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
-                                                               *winptr--, right_shifts);
-    }
-}
-
-void WebRtcSpl_ElementwiseVectorMult(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in,
-                                     G_CONST WebRtc_Word16 *win, WebRtc_Word16 vector_length,
-                                     WebRtc_Word16 right_shifts)
-{
-    int i;
-    WebRtc_Word16 *outptr = out;
-    G_CONST WebRtc_Word16 *inptr = in;
-    G_CONST WebRtc_Word16 *winptr = win;
-    for (i = 0; i < vector_length; i++)
-    {
-        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
-                                                               *winptr++, right_shifts);
-    }
-}
-
-void WebRtcSpl_AddVectorsAndShift(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in1,
-                                  G_CONST WebRtc_Word16 *in2, WebRtc_Word16 vector_length,
-                                  WebRtc_Word16 right_shifts)
-{
-    int i;
-    WebRtc_Word16 *outptr = out;
-    G_CONST WebRtc_Word16 *in1ptr = in1;
-    G_CONST WebRtc_Word16 *in2ptr = in2;
-    for (i = vector_length; i > 0; i--)
-    {
-        (*outptr++) = (WebRtc_Word16)(((*in1ptr++) + (*in2ptr++)) >> right_shifts);
-    }
-}
-
-void WebRtcSpl_AddAffineVectorToVector(WebRtc_Word16 *out, WebRtc_Word16 *in,
-                                       WebRtc_Word16 gain, WebRtc_Word32 add_constant,
-                                       WebRtc_Word16 right_shifts, int vector_length)
-{
-    WebRtc_Word16 *inPtr;
-    WebRtc_Word16 *outPtr;
-    int i;
-
-    inPtr = in;
-    outPtr = out;
-    for (i = 0; i < vector_length; i++)
-    {
-        (*outPtr++) += (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
-                + (WebRtc_Word32)add_constant) >> right_shifts);
-    }
-}
-
-void WebRtcSpl_AffineTransformVector(WebRtc_Word16 *out, WebRtc_Word16 *in,
-                                     WebRtc_Word16 gain, WebRtc_Word32 add_constant,
-                                     WebRtc_Word16 right_shifts, int vector_length)
-{
-    WebRtc_Word16 *inPtr;
-    WebRtc_Word16 *outPtr;
-    int i;
-
-    inPtr = in;
-    outPtr = out;
-    for (i = 0; i < vector_length; i++)
-    {
-        (*outPtr++) = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
-                + (WebRtc_Word32)add_constant) >> right_shifts);
-    }
-}

src/common_audio/signal_processing_library/main/source/levinson_durbin.c

@@ -1,259 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_LevinsonDurbin().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define SPL_LEVINSON_MAXORDER 20
-
-WebRtc_Word16 WebRtcSpl_LevinsonDurbin(WebRtc_Word32 *R, WebRtc_Word16 *A, WebRtc_Word16 *K,
-                                       WebRtc_Word16 order)
-{
-    WebRtc_Word16 i, j;
-    // Auto-correlation coefficients in high precision
-    WebRtc_Word16 R_hi[SPL_LEVINSON_MAXORDER + 1], R_low[SPL_LEVINSON_MAXORDER + 1];
-    // LPC coefficients in high precision
-    WebRtc_Word16 A_hi[SPL_LEVINSON_MAXORDER + 1], A_low[SPL_LEVINSON_MAXORDER + 1];
-    // LPC coefficients for next iteration
-    WebRtc_Word16 A_upd_hi[SPL_LEVINSON_MAXORDER + 1], A_upd_low[SPL_LEVINSON_MAXORDER + 1];
-    // Reflection coefficient in high precision
-    WebRtc_Word16 K_hi, K_low;
-    // Prediction gain Alpha in high precision and with scale factor
-    WebRtc_Word16 Alpha_hi, Alpha_low, Alpha_exp;
-    WebRtc_Word16 tmp_hi, tmp_low;
-    WebRtc_Word32 temp1W32, temp2W32, temp3W32;
-    WebRtc_Word16 norm;
-
-    // Normalize the autocorrelation R[0]...R[order+1]
-
-    norm = WebRtcSpl_NormW32(R[0]);
-
-    for (i = order; i >= 0; i--)
-    {
-        temp1W32 = WEBRTC_SPL_LSHIFT_W32(R[i], norm);
-        // Put R in hi and low format
-        R_hi[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-        R_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)), 1);
-    }
-
-    // K = A[1] = -R[1] / R[0]
-
-    temp2W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[1],16)
-            + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[1],1); // R[1] in Q31
-    temp3W32 = WEBRTC_SPL_ABS_W32(temp2W32); // abs R[1]
-    temp1W32 = WebRtcSpl_DivW32HiLow(temp3W32, R_hi[0], R_low[0]); // abs(R[1])/R[0] in Q31
-    // Put back the sign on R[1]
-    if (temp2W32 > 0)
-    {
-        temp1W32 = -temp1W32;
-    }
-
-    // Put K in hi and low format
-    K_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-    K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
-
-    // Store first reflection coefficient
-    K[0] = K_hi;
-
-    temp1W32 = WEBRTC_SPL_RSHIFT_W32(temp1W32, 4); // A[1] in Q27
-
-    // Put A[1] in hi and low format
-    A_hi[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-    A_low[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[1], 16)), 1);
-
-    // Alpha = R[0] * (1-K^2)
-
-    temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14) + WEBRTC_SPL_MUL_16_16(K_hi, K_hi))
-            << 1); // temp1W32 = k^2 in Q31
-
-    temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
-    temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; // temp1W32 = (1 - K[0]*K[0]) in Q31
-
-    // Store temp1W32 = 1 - K[0]*K[0] on hi and low format
-    tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-    tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
-    // Calculate Alpha in Q31
-    temp1W32 = ((WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_hi)
-            + (WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_low) >> 15)
-            + (WEBRTC_SPL_MUL_16_16(R_low[0], tmp_hi) >> 15)) << 1);
-
-    // Normalize Alpha and put it in hi and low format
-
-    Alpha_exp = WebRtcSpl_NormW32(temp1W32);
-    temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, Alpha_exp);
-    Alpha_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-    Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
-
-    // Perform the iterative calculations in the Levinson-Durbin algorithm
-
-    for (i = 2; i <= order; i++)
-    {
-        /*                    ----
-         temp1W32 =  R[i] + > R[j]*A[i-j]
-         /
-         ----
-         j=1..i-1
-         */
-
-        temp1W32 = 0;
-
-        for (j = 1; j < i; j++)
-        {
-            // temp1W32 is in Q31
-            temp1W32 += ((WEBRTC_SPL_MUL_16_16(R_hi[j], A_hi[i-j]) << 1)
-                    + (((WEBRTC_SPL_MUL_16_16(R_hi[j], A_low[i-j]) >> 15)
-                            + (WEBRTC_SPL_MUL_16_16(R_low[j], A_hi[i-j]) >> 15)) << 1));
-        }
-
-        temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4);
-        temp1W32 += (WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)
-                + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[i], 1));
-
-        // K = -temp1W32 / Alpha
-        temp2W32 = WEBRTC_SPL_ABS_W32(temp1W32); // abs(temp1W32)
-        temp3W32 = WebRtcSpl_DivW32HiLow(temp2W32, Alpha_hi, Alpha_low); // abs(temp1W32)/Alpha
-
-        // Put the sign of temp1W32 back again
-        if (temp1W32 > 0)
-        {
-            temp3W32 = -temp3W32;
-        }
-
-        // Use the Alpha shifts from earlier to de-normalize
-        norm = WebRtcSpl_NormW32(temp3W32);
-        if ((Alpha_exp <= norm) || (temp3W32 == 0))
-        {
-            temp3W32 = WEBRTC_SPL_LSHIFT_W32(temp3W32, Alpha_exp);
-        } else
-        {
-            if (temp3W32 > 0)
-            {
-                temp3W32 = (WebRtc_Word32)0x7fffffffL;
-            } else
-            {
-                temp3W32 = (WebRtc_Word32)0x80000000L;
-            }
-        }
-
-        // Put K on hi and low format
-        K_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
-        K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32
-                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
-
-        // Store Reflection coefficient in Q15
-        K[i - 1] = K_hi;
-
-        // Test for unstable filter.
-        // If unstable return 0 and let the user decide what to do in that case
-
-        if ((WebRtc_Word32)WEBRTC_SPL_ABS_W16(K_hi) > (WebRtc_Word32)32750)
-        {
-            return 0; // Unstable filter
-        }
-
-        /*
-         Compute updated LPC coefficient: Anew[i]
-         Anew[j]= A[j] + K*A[i-j]   for j=1..i-1
-         Anew[i]= K
-         */
-
-        for (j = 1; j < i; j++)
-        {
-            // temp1W32 = A[j] in Q27
-            temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[j],16)
-                    + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[j],1);
-
-            // temp1W32 += K*A[i-j] in Q27
-            temp1W32 += ((WEBRTC_SPL_MUL_16_16(K_hi, A_hi[i-j])
-                    + (WEBRTC_SPL_MUL_16_16(K_hi, A_low[i-j]) >> 15)
-                    + (WEBRTC_SPL_MUL_16_16(K_low, A_hi[i-j]) >> 15)) << 1);
-
-            // Put Anew in hi and low format
-            A_upd_hi[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-            A_upd_low[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-                    - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[j], 16)), 1);
-        }
-
-        // temp3W32 = K in Q27 (Convert from Q31 to Q27)
-        temp3W32 = WEBRTC_SPL_RSHIFT_W32(temp3W32, 4);
-
-        // Store Anew in hi and low format
-        A_upd_hi[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
-        A_upd_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32
-                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[i], 16)), 1);
-
-        // Alpha = Alpha * (1-K^2)
-
-        temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14)
-                + WEBRTC_SPL_MUL_16_16(K_hi, K_hi)) << 1); // K*K in Q31
-
-        temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
-        temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; // 1 - K*K  in Q31
-
-        // Convert 1- K^2 in hi and low format
-        tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-        tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
-        // Calculate Alpha = Alpha * (1-K^2) in Q31
-        temp1W32 = ((WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_hi)
-                + (WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_low) >> 15)
-                + (WEBRTC_SPL_MUL_16_16(Alpha_low, tmp_hi) >> 15)) << 1);
-
-        // Normalize Alpha and store it on hi and low format
-
-        norm = WebRtcSpl_NormW32(temp1W32);
-        temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, norm);
-
-        Alpha_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-        Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
-
-        // Update the total normalization of Alpha
-        Alpha_exp = Alpha_exp + norm;
-
-        // Update A[]
-
-        for (j = 1; j <= i; j++)
-        {
-            A_hi[j] = A_upd_hi[j];
-            A_low[j] = A_upd_low[j];
-        }
-    }
-
-    /*
-     Set A[0] to 1.0 and store the A[i] i=1...order in Q12
-     (Convert from Q27 and use rounding)
-     */
-
-    A[0] = 4096;
-
-    for (i = 1; i <= order; i++)
-    {
-        // temp1W32 in Q27
-        temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[i], 16)
-                + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[i], 1);
-        // Round and store upper word
-        A[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32<<1)+(WebRtc_Word32)32768, 16);
-    }
-    return 1; // Stable filters
-}

src/common_audio/signal_processing_library/main/source/min_max_operations.c

@@ -1,265 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the implementation of functions
- * WebRtcSpl_MaxAbsValueW16()
- * WebRtcSpl_MaxAbsIndexW16()
- * WebRtcSpl_MaxAbsValueW32()
- * WebRtcSpl_MaxValueW16()
- * WebRtcSpl_MaxIndexW16()
- * WebRtcSpl_MaxValueW32()
- * WebRtcSpl_MaxIndexW32()
- * WebRtcSpl_MinValueW16()
- * WebRtcSpl_MinIndexW16()
- * WebRtcSpl_MinValueW32()
- * WebRtcSpl_MinIndexW32()
- *
- * The description header can be found in signal_processing_library.h.
- *
- */
-
-#include "signal_processing_library.h"
-
-#if !(defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
-
-// Maximum absolute value of word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word32 tempMax = 0;
-    WebRtc_Word32 absVal;
-    WebRtc_Word16 totMax;
-    int i;
-    G_CONST WebRtc_Word16 *tmpvector = vector;
-
-    for (i = 0; i < length; i++)
-    {
-        absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
-        if (absVal > tempMax)
-        {
-            tempMax = absVal;
-        }
-        tmpvector++;
-    }
-    totMax = (WebRtc_Word16)WEBRTC_SPL_MIN(tempMax, WEBRTC_SPL_WORD16_MAX);
-    return totMax;
-}
-
-#endif
-
-// Index of maximum absolute value in a  word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxAbsIndexW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 tempMax;
-    WebRtc_Word16 absTemp;
-    WebRtc_Word16 tempMaxIndex = 0;
-    WebRtc_Word16 i = 0;
-    G_CONST WebRtc_Word16 *tmpvector = vector;
-
-    tempMax = WEBRTC_SPL_ABS_W16(*tmpvector);
-    tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        absTemp = WEBRTC_SPL_ABS_W16(*tmpvector);
-        tmpvector++;
-        if (absTemp > tempMax)
-        {
-            tempMax = absTemp;
-            tempMaxIndex = i;
-        }
-    }
-    return tempMaxIndex;
-}
-
-// Maximum absolute value of word32 vector.
-WebRtc_Word32 WebRtcSpl_MaxAbsValueW32(G_CONST WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
-    WebRtc_UWord32 tempMax = 0;
-    WebRtc_UWord32 absVal;
-    WebRtc_Word32 retval;
-    int i;
-    G_CONST WebRtc_Word32 *tmpvector = vector;
-
-    for (i = 0; i < length; i++)
-    {
-        absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
-        if (absVal > tempMax)
-        {
-            tempMax = absVal;
-        }
-        tmpvector++;
-    }
-    retval = (WebRtc_Word32)(WEBRTC_SPL_MIN(tempMax, WEBRTC_SPL_WORD32_MAX));
-    return retval;
-}
-
-// Maximum value of word16 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word16 WebRtcSpl_MaxValueW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 tempMax;
-    WebRtc_Word16 i;
-    G_CONST WebRtc_Word16 *tmpvector = vector;
-
-    tempMax = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ > tempMax)
-            tempMax = vector[i];
-    }
-    return tempMax;
-}
-#else
-#pragma message(">> WebRtcSpl_MaxValueW16 is excluded from this build")
-#endif
-
-// Index of maximum value in a word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxIndexW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 tempMax;
-    WebRtc_Word16 tempMaxIndex = 0;
-    WebRtc_Word16 i = 0;
-    G_CONST WebRtc_Word16 *tmpvector = vector;
-
-    tempMax = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ > tempMax)
-        {
-            tempMax = vector[i];
-            tempMaxIndex = i;
-        }
-    }
-    return tempMaxIndex;
-}
-
-// Maximum value of word32 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word32 WebRtcSpl_MaxValueW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word32 tempMax;
-    WebRtc_Word16 i;
-    G_CONST WebRtc_Word32 *tmpvector = vector;
-
-    tempMax = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ > tempMax)
-            tempMax = vector[i];
-    }
-    return tempMax;
-}
-#else
-#pragma message(">> WebRtcSpl_MaxValueW32 is excluded from this build")
-#endif
-
-// Index of maximum value in a word32 vector.
-WebRtc_Word16 WebRtcSpl_MaxIndexW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word32 tempMax;
-    WebRtc_Word16 tempMaxIndex = 0;
-    WebRtc_Word16 i = 0;
-    G_CONST WebRtc_Word32 *tmpvector = vector;
-
-    tempMax = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ > tempMax)
-        {
-            tempMax = vector[i];
-            tempMaxIndex = i;
-        }
-    }
-    return tempMaxIndex;
-}
-
-// Minimum value of word16 vector.
-WebRtc_Word16 WebRtcSpl_MinValueW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 tempMin;
-    WebRtc_Word16 i;
-    G_CONST WebRtc_Word16 *tmpvector = vector;
-
-    // Find the minimum value
-    tempMin = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ < tempMin)
-            tempMin = (vector[i]);
-    }
-    return tempMin;
-}
-
-// Index of minimum value in a word16 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word16 WebRtcSpl_MinIndexW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 tempMin;
-    WebRtc_Word16 tempMinIndex = 0;
-    WebRtc_Word16 i = 0;
-    G_CONST WebRtc_Word16* tmpvector = vector;
-
-    // Find index of smallest value
-    tempMin = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ < tempMin)
-        {
-            tempMin = vector[i];
-            tempMinIndex = i;
-        }
-    }
-    return tempMinIndex;
-}
-#else
-#pragma message(">> WebRtcSpl_MinIndexW16 is excluded from this build")
-#endif
-
-// Minimum value of word32 vector.
-WebRtc_Word32 WebRtcSpl_MinValueW32(G_CONST WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word32 tempMin;
-    WebRtc_Word16 i;
-    G_CONST WebRtc_Word32 *tmpvector = vector;
-
-    // Find the minimum value
-    tempMin = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ < tempMin)
-            tempMin = (vector[i]);
-    }
-    return tempMin;
-}
-
-// Index of minimum value in a word32 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word16 WebRtcSpl_MinIndexW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word32 tempMin;
-    WebRtc_Word16 tempMinIndex = 0;
-    WebRtc_Word16 i = 0;
-    G_CONST WebRtc_Word32 *tmpvector = vector;
-
-    // Find index of smallest value
-    tempMin = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ < tempMin)
-        {
-            tempMin = vector[i];
-            tempMinIndex = i;
-        }
-    }
-    return tempMinIndex;
-}
-#else
-#pragma message(">> WebRtcSpl_MinIndexW32 is excluded from this build")
-#endif

src/common_audio/signal_processing_library/main/source/min_max_operations_neon.c

@@ -1,47 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#if (defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
-
-#include <arm_neon.h>
-
-#include "signal_processing_library.h"
-
-// Maximum absolute value of word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16* vector,
-                                       WebRtc_Word16 length) {
-  WebRtc_Word32 temp_max = 0;
-  WebRtc_Word32 abs_val;
-  WebRtc_Word16 tot_max;
-  int i;
-
-  __asm__("vmov.i16 d25, #0" : : : "d25");
-
-  for (i = 0; i < length - 7; i += 8) {
-    __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&vector[i]) : "q13");
-    __asm__("vabs.s16 q13, q13" : : : "q13");
-    __asm__("vpmax.s16 d26, d27" : : : "q13");
-    __asm__("vpmax.s16 d25, d26" : : : "d25", "d26");
-  }
-  __asm__("vpmax.s16 d25, d25" : : : "d25");
-  __asm__("vpmax.s16 d25, d25" : : : "d25");
-  __asm__("vmov.s16 %0, d25[0]" : "=r"(temp_max): : "d25");
-
-  for (; i < length; i++) {
-    abs_val = WEBRTC_SPL_ABS_W32((vector[i]));
-    if (abs_val > temp_max) {
-      temp_max = abs_val;
-    }
-  }
-  tot_max = (WebRtc_Word16)WEBRTC_SPL_MIN(temp_max, WEBRTC_SPL_WORD16_MAX);
-  return tot_max;
-}
-
-#endif

src/common_audio/signal_processing_library/main/source/randomization_functions.c

@@ -1,52 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the randomization functions
- * WebRtcSpl_IncreaseSeed()
- * WebRtcSpl_RandU()
- * WebRtcSpl_RandN()
- * WebRtcSpl_RandUArray()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32 *seed)
-{
-    seed[0] = (seed[0] * ((WebRtc_Word32)69069) + 1) & (WEBRTC_SPL_MAX_SEED_USED - 1);
-    return seed[0];
-}
-
-WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32 *seed)
-{
-    return (WebRtc_Word16)(WebRtcSpl_IncreaseSeed(seed) >> 16);
-}
-
-WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32 *seed)
-{
-    return WebRtcSpl_kRandNTable[WebRtcSpl_IncreaseSeed(seed) >> 23];
-}
-
-// Creates an array of uniformly distributed variables
-WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
-                                   WebRtc_Word16 vector_length,
-                                   WebRtc_UWord32* seed)
-{
-    int i;
-    for (i = 0; i < vector_length; i++)
-    {
-        vector[i] = WebRtcSpl_RandU(seed);
-    }
-    return vector_length;
-}

src/common_audio/signal_processing_library/main/source/resample_by_2_internal.h

@@ -1,47 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file contains some internal resampling functions.
- *
- */
-
-#ifndef WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
-#define WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
-
-#include "typedefs.h"
-
-/*******************************************************************
- * resample_by_2_fast.c
- * Functions for internal use in the other resample functions
- ******************************************************************/
-void WebRtcSpl_DownBy2IntToShort(WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
-                                 WebRtc_Word32 *state);
-
-void WebRtcSpl_DownBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len,
-                                 WebRtc_Word32 *out, WebRtc_Word32 *state);
-
-void WebRtcSpl_UpBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len,
-                               WebRtc_Word32 *out, WebRtc_Word32 *state);
-
-void WebRtcSpl_UpBy2IntToInt(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
-                             WebRtc_Word32 *state);
-
-void WebRtcSpl_UpBy2IntToShort(const WebRtc_Word32 *in, WebRtc_Word32 len,
-                               WebRtc_Word16 *out, WebRtc_Word32 *state);
-
-void WebRtcSpl_LPBy2ShortToInt(const WebRtc_Word16* in, WebRtc_Word32 len,
-                               WebRtc_Word32* out, WebRtc_Word32* state);
-
-void WebRtcSpl_LPBy2IntToInt(const WebRtc_Word32* in, WebRtc_Word32 len, WebRtc_Word32* out,
-                             WebRtc_Word32* state);
-
-#endif // WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_

src/common_audio/signal_processing_library/main/source/sin_table.c

@@ -1,60 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the 360 degree sine table.
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_kSinTable[] = {
-        0,    142,    285,    428,    571,    713,    856,    998,   1140,
-     1281,   1422,   1563,   1703,   1842,   1981,   2120,   2258,   2395,
-     2531,   2667,   2801,   2935,   3068,   3200,   3331,   3462,   3591,
-     3719,   3845,   3971,   4095,   4219,   4341,   4461,   4580,   4698,
-     4815,   4930,   5043,   5155,   5265,   5374,   5481,   5586,   5690,
-     5792,   5892,   5991,   6087,   6182,   6275,   6366,   6455,   6542,
-     6627,   6710,   6791,   6870,   6947,   7021,   7094,   7164,   7233,
-     7299,   7362,   7424,   7483,   7540,   7595,   7647,   7697,   7745,
-     7791,   7834,   7874,   7912,   7948,   7982,   8012,   8041,   8067,
-     8091,   8112,   8130,   8147,   8160,   8172,   8180,   8187,   8190,
-     8191,   8190,   8187,   8180,   8172,   8160,   8147,   8130,   8112,
-     8091,   8067,   8041,   8012,   7982,   7948,   7912,   7874,   7834,
-     7791,   7745,   7697,   7647,   7595,   7540,   7483,   7424,   7362,
-     7299,   7233,   7164,   7094,   7021,   6947,   6870,   6791,   6710,
-     6627,   6542,   6455,   6366,   6275,   6182,   6087,   5991,   5892,
-     5792,   5690,   5586,   5481,   5374,   5265,   5155,   5043,   4930,
-     4815,   4698,   4580,   4461,   4341,   4219,   4096,   3971,   3845,
-     3719,   3591,   3462,   3331,   3200,   3068,   2935,   2801,   2667,
-     2531,   2395,   2258,   2120,   1981,   1842,   1703,   1563,   1422,
-     1281,   1140,    998,    856,    713,    571,    428,    285,    142,
-        0,   -142,   -285,   -428,   -571,   -713,   -856,   -998,  -1140,
-    -1281,  -1422,  -1563,  -1703,  -1842,  -1981,  -2120,  -2258,  -2395,
-    -2531,  -2667,  -2801,  -2935,  -3068,  -3200,  -3331,  -3462,  -3591,
-    -3719,  -3845,  -3971,  -4095,  -4219,  -4341,  -4461,  -4580,  -4698,
-    -4815,  -4930,  -5043,  -5155,  -5265,  -5374,  -5481,  -5586,  -5690,
-    -5792,  -5892,  -5991,  -6087,  -6182,  -6275,  -6366,  -6455,  -6542,
-    -6627,  -6710,  -6791,  -6870,  -6947,  -7021,  -7094,  -7164,  -7233,
-    -7299,  -7362,  -7424,  -7483,  -7540,  -7595,  -7647,  -7697,  -7745,
-    -7791,  -7834,  -7874,  -7912,  -7948,  -7982,  -8012,  -8041,  -8067,
-    -8091,  -8112,  -8130,  -8147,  -8160,  -8172,  -8180,  -8187,  -8190,
-    -8191,  -8190,  -8187,  -8180,  -8172,  -8160,  -8147,  -8130,  -8112,
-    -8091,  -8067,  -8041,  -8012,  -7982,  -7948,  -7912,  -7874,  -7834,
-    -7791,  -7745,  -7697,  -7647,  -7595,  -7540,  -7483,  -7424,  -7362,
-    -7299,  -7233,  -7164,  -7094,  -7021,  -6947,  -6870,  -6791,  -6710,
-    -6627,  -6542,  -6455,  -6366,  -6275,  -6182,  -6087,  -5991,  -5892,
-    -5792,  -5690,  -5586,  -5481,  -5374,  -5265,  -5155,  -5043,  -4930,
-    -4815,  -4698,  -4580,  -4461,  -4341,  -4219,  -4096,  -3971,  -3845,
-    -3719,  -3591,  -3462,  -3331,  -3200,  -3068,  -2935,  -2801,  -2667,
-    -2531,  -2395,  -2258,  -2120,  -1981,  -1842,  -1703,  -1563,  -1422,
-    -1281,  -1140,   -998,   -856,   -713,   -571,   -428,   -285,   -142
-};

src/common_audio/signal_processing_library/main/source/sin_table_1024.c

@@ -1,150 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the 1024 point sine table.
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_kSinTable1024[] =
-{
-      0,    201,    402,    603,    804,   1005,   1206,   1406,
-   1607,   1808,   2009,   2209,   2410,   2610,   2811,   3011,
-   3211,   3411,   3611,   3811,   4011,   4210,   4409,   4608,
-   4807,   5006,   5205,   5403,   5601,   5799,   5997,   6195,
-   6392,   6589,   6786,   6982,   7179,   7375,   7571,   7766,
-   7961,   8156,   8351,   8545,   8739,   8932,   9126,   9319,
-   9511,   9703,   9895,  10087,  10278,  10469,  10659,  10849,
-  11038,  11227,  11416,  11604,  11792,  11980,  12166,  12353,
-  12539,  12724,  12909,  13094,  13278,  13462,  13645,  13827,
-  14009,  14191,  14372,  14552,  14732,  14911,  15090,  15268,
-  15446,  15623,  15799,  15975,  16150,  16325,  16499,  16672,
-  16845,  17017,  17189,  17360,  17530,  17699,  17868,  18036,
-  18204,  18371,  18537,  18702,  18867,  19031,  19194,  19357,
-  19519,  19680,  19840,  20000,  20159,  20317,  20474,  20631,
-  20787,  20942,  21096,  21249,  21402,  21554,  21705,  21855,
-  22004,  22153,  22301,  22448,  22594,  22739,  22883,  23027,
-  23169,  23311,  23452,  23592,  23731,  23869,  24006,  24143,
-  24278,  24413,  24546,  24679,  24811,  24942,  25072,  25201,
-  25329,  25456,  25582,  25707,  25831,  25954,  26077,  26198,
-  26318,  26437,  26556,  26673,  26789,  26905,  27019,  27132,
-  27244,  27355,  27466,  27575,  27683,  27790,  27896,  28001,
-  28105,  28208,  28309,  28410,  28510,  28608,  28706,  28802,
-  28897,  28992,  29085,  29177,  29268,  29358,  29446,  29534,
-  29621,  29706,  29790,  29873,  29955,  30036,  30116,  30195,
-  30272,  30349,  30424,  30498,  30571,  30643,  30713,  30783,
-  30851,  30918,  30984,  31049,
-  31113,  31175,  31236,  31297,
-  31356,  31413,  31470,  31525,  31580,  31633,  31684,  31735,
-  31785,  31833,  31880,  31926,  31970,  32014,  32056,  32097,
-  32137,  32176,  32213,  32249,  32284,  32318,  32350,  32382,
-  32412,  32441,  32468,  32495,  32520,  32544,  32567,  32588,
-  32609,  32628,  32646,  32662,  32678,  32692,  32705,  32717,
-  32727,  32736,  32744,  32751,  32757,  32761,  32764,  32766,
-  32767,  32766,  32764,  32761,  32757,  32751,  32744,  32736,
-  32727,  32717,  32705,  32692,  32678,  32662,  32646,  32628,
-  32609,  32588,  32567,  32544,  32520,  32495,  32468,  32441,
-  32412,  32382,  32350,  32318,  32284,  32249,  32213,  32176,
-  32137,  32097,  32056,  32014,  31970,  31926,  31880,  31833,
-  31785,  31735,  31684,  31633,  31580,  31525,  31470,  31413,
-  31356,  31297,  31236,  31175,  31113,  31049,  30984,  30918,
-  30851,  30783,  30713,  30643,  30571,  30498,  30424,  30349,
-  30272,  30195,  30116,  30036,  29955,  29873,  29790,  29706,
-  29621,  29534,  29446,  29358,  29268,  29177,  29085,  28992,
-  28897,  28802,  28706,  28608,  28510,  28410,  28309,  28208,
-  28105,  28001,  27896,  27790,  27683,  27575,  27466,  27355,
-  27244,  27132,  27019,  26905,  26789,  26673,  26556,  26437,
-  26318,  26198,  26077,  25954,  25831,  25707,  25582,  25456,
-  25329,  25201,  25072,  24942,  24811,  24679,  24546,  24413,
-  24278,  24143,  24006,  23869,  23731,  23592,  23452,  23311,
-  23169,  23027,  22883,  22739,  22594,  22448,  22301,  22153,
-  22004,  21855,  21705,  21554,  21402,  21249,  21096,  20942,
-  20787,  20631,  20474,  20317,  20159,  20000,  19840,  19680,
-  19519,  19357,  19194,  19031,  18867,  18702,  18537,  18371,
-  18204,  18036,  17868,  17699,  17530,  17360,  17189,  17017,
-  16845,  16672,  16499,  16325,  16150,  15975,  15799,  15623,
-  15446,  15268,  15090,  14911,  14732,  14552,  14372,  14191,
-  14009,  13827,  13645,  13462,  13278,  13094,  12909,  12724,
-  12539,  12353,  12166,  11980,  11792,  11604,  11416,  11227,
-  11038,  10849,  10659,  10469,  10278,  10087,   9895,   9703,
-   9511,   9319,   9126,   8932,   8739,   8545,   8351,   8156,
-   7961,   7766,   7571,   7375,   7179,   6982,   6786,   6589,
-   6392,   6195,   5997,   5799,   5601,   5403,   5205,   5006,
-   4807,   4608,   4409,   4210,   4011,   3811,   3611,   3411,
-   3211,   3011,   2811,   2610,   2410,   2209,   2009,   1808,
-   1607,   1406,   1206,   1005,    804,    603,    402,    201,
-      0,   -201,   -402,   -603,   -804,  -1005,  -1206,  -1406,
-  -1607,  -1808,  -2009,  -2209,  -2410,  -2610,  -2811,  -3011,
-  -3211,  -3411,  -3611,  -3811,  -4011,  -4210,  -4409,  -4608,
-  -4807,  -5006,  -5205,  -5403,  -5601,  -5799,  -5997,  -6195,
-  -6392,  -6589,  -6786,  -6982,  -7179,  -7375,  -7571,  -7766,
-  -7961,  -8156,  -8351,  -8545,  -8739,  -8932,  -9126,  -9319,
-  -9511,  -9703,  -9895, -10087, -10278, -10469, -10659, -10849,
- -11038, -11227, -11416, -11604, -11792, -11980, -12166, -12353,
- -12539, -12724, -12909, -13094, -13278, -13462, -13645, -13827,
- -14009, -14191, -14372, -14552, -14732, -14911, -15090, -15268,
- -15446, -15623, -15799, -15975, -16150, -16325, -16499, -16672,
- -16845, -17017, -17189, -17360, -17530, -17699, -17868, -18036,
- -18204, -18371, -18537, -18702, -18867, -19031, -19194, -19357,
- -19519, -19680, -19840, -20000, -20159, -20317, -20474, -20631,
- -20787, -20942, -21096, -21249, -21402, -21554, -21705, -21855,
- -22004, -22153, -22301, -22448, -22594, -22739, -22883, -23027,
- -23169, -23311, -23452, -23592, -23731, -23869, -24006, -24143,
- -24278, -24413, -24546, -24679, -24811, -24942, -25072, -25201,
- -25329, -25456, -25582, -25707, -25831, -25954, -26077, -26198,
- -26318, -26437, -26556, -26673, -26789, -26905, -27019, -27132,
- -27244, -27355, -27466, -27575, -27683, -27790, -27896, -28001,
- -28105, -28208, -28309, -28410, -28510, -28608, -28706, -28802,
- -28897, -28992, -29085, -29177, -29268, -29358, -29446, -29534,
- -29621, -29706, -29790, -29873, -29955, -30036, -30116, -30195,
- -30272, -30349, -30424, -30498, -30571, -30643, -30713, -30783,
- -30851, -30918, -30984, -31049, -31113, -31175, -31236, -31297,
- -31356, -31413, -31470, -31525, -31580, -31633, -31684, -31735,
- -31785, -31833, -31880, -31926, -31970, -32014, -32056, -32097,
- -32137, -32176, -32213, -32249, -32284, -32318, -32350, -32382,
- -32412, -32441, -32468, -32495, -32520, -32544, -32567, -32588,
- -32609, -32628, -32646, -32662, -32678, -32692, -32705, -32717,
- -32727, -32736, -32744, -32751, -32757, -32761, -32764, -32766,
- -32767, -32766, -32764, -32761, -32757, -32751, -32744, -32736,
- -32727, -32717, -32705, -32692, -32678, -32662, -32646, -32628,
- -32609, -32588, -32567, -32544, -32520, -32495, -32468, -32441,
- -32412, -32382, -32350, -32318, -32284, -32249, -32213, -32176,
- -32137, -32097, -32056, -32014, -31970, -31926, -31880, -31833,
- -31785, -31735, -31684, -31633, -31580, -31525, -31470, -31413,
- -31356, -31297, -31236, -31175, -31113, -31049, -30984, -30918,
- -30851, -30783, -30713, -30643, -30571, -30498, -30424, -30349,
- -30272, -30195, -30116, -30036, -29955, -29873, -29790, -29706,
- -29621, -29534, -29446, -29358, -29268, -29177, -29085, -28992,
- -28897, -28802, -28706, -28608, -28510, -28410, -28309, -28208,
- -28105, -28001, -27896, -27790, -27683, -27575, -27466, -27355,
- -27244, -27132, -27019, -26905, -26789, -26673, -26556, -26437,
- -26318, -26198, -26077, -25954, -25831, -25707, -25582, -25456,
- -25329, -25201, -25072, -24942, -24811, -24679, -24546, -24413,
- -24278, -24143, -24006, -23869, -23731, -23592, -23452, -23311,
- -23169, -23027, -22883, -22739, -22594, -22448, -22301, -22153,
- -22004, -21855, -21705, -21554, -21402, -21249, -21096, -20942,
- -20787, -20631, -20474, -20317, -20159, -20000, -19840, -19680,
- -19519, -19357, -19194, -19031, -18867, -18702, -18537, -18371,
- -18204, -18036, -17868, -17699, -17530, -17360, -17189, -17017,
- -16845, -16672, -16499, -16325, -16150, -15975, -15799, -15623,
- -15446, -15268, -15090, -14911, -14732, -14552, -14372, -14191,
- -14009, -13827, -13645, -13462, -13278, -13094, -12909, -12724,
- -12539, -12353, -12166, -11980, -11792, -11604, -11416, -11227,
- -11038, -10849, -10659, -10469, -10278, -10087,  -9895,  -9703,
-  -9511,  -9319,  -9126,  -8932,  -8739,  -8545,  -8351,  -8156,
-  -7961,  -7766,  -7571,  -7375,  -7179,  -6982,  -6786,  -6589,
-  -6392,  -6195,  -5997,  -5799,  -5601,  -5403,  -5205,  -5006,
-  -4807,  -4608,  -4409,  -4210,  -4011,  -3811,  -3611,  -3411,
-  -3211,  -3011,  -2811,  -2610,  -2410,  -2209,  -2009,  -1808,
-  -1607,  -1406,  -1206,  -1005,   -804,   -603,   -402,   -201,
-};

src/common_audio/signal_processing_library/main/source/spl.gypi

@@ -1,73 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'targets': [
-    {
-      'target_name': 'spl',
-      'type': '<(library)',
-      'include_dirs': [
-        '../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-        ],
-      },
-      'sources': [
-        '../interface/signal_processing_library.h',
-        '../interface/spl_inl.h',
-        'auto_corr_to_refl_coef.c',
-        'auto_correlation.c',
-        'complex_fft.c',
-        'complex_ifft.c',
-        'complex_bit_reverse.c',
-        'copy_set_operations.c',
-        'cos_table.c',
-        'cross_correlation.c',
-        'division_operations.c',
-        'dot_product_with_scale.c',
-        'downsample_fast.c',
-        'energy.c',
-        'filter_ar.c',
-        'filter_ar_fast_q12.c',
-        'filter_ma_fast_q12.c',
-        'get_hanning_window.c',
-        'get_scaling_square.c',
-        'hanning_table.c',
-        'ilbc_specific_functions.c',
-        'levinson_durbin.c',
-        'lpc_to_refl_coef.c',
-        'min_max_operations.c',
-        'randn_table.c',
-        'randomization_functions.c',
-        'refl_coef_to_lpc.c',
-        'resample.c',
-        'resample_48khz.c',
-        'resample_by_2.c',
-        'resample_by_2_internal.c',
-        'resample_by_2_internal.h',
-        'resample_fractional.c',
-        'sin_table.c',
-        'sin_table_1024.c',
-        'spl_sqrt.c',
-        'spl_sqrt_floor.c',
-        'spl_version.c',
-        'splitting_filter.c',
-        'sqrt_of_one_minus_x_squared.c',
-        'vector_scaling_operations.c',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:

src/common_audio/signal_processing_library/main/source/spl_sqrt_floor.c

@@ -1,53 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the function WebRtcSpl_SqrtFloor().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define WEBRTC_SPL_SQRT_ITER(N)                 \
-  try1 = root + (1 << (N));                     \
-  if (value >= try1 << (N))                     \
-  {                                             \
-    value -= try1 << (N);                       \
-    root |= 2 << (N);                           \
-  }
-
-// (out) Square root of input parameter
-WebRtc_Word32 WebRtcSpl_SqrtFloor(WebRtc_Word32 value)
-{
-    // new routine for performance, 4 cycles/bit in ARM
-    // output precision is 16 bits
-
-    WebRtc_Word32 root = 0, try1;
-
-    WEBRTC_SPL_SQRT_ITER (15);
-    WEBRTC_SPL_SQRT_ITER (14);
-    WEBRTC_SPL_SQRT_ITER (13);
-    WEBRTC_SPL_SQRT_ITER (12);
-    WEBRTC_SPL_SQRT_ITER (11);
-    WEBRTC_SPL_SQRT_ITER (10);
-    WEBRTC_SPL_SQRT_ITER ( 9);
-    WEBRTC_SPL_SQRT_ITER ( 8);
-    WEBRTC_SPL_SQRT_ITER ( 7);
-    WEBRTC_SPL_SQRT_ITER ( 6);
-    WEBRTC_SPL_SQRT_ITER ( 5);
-    WEBRTC_SPL_SQRT_ITER ( 4);
-    WEBRTC_SPL_SQRT_ITER ( 3);
-    WEBRTC_SPL_SQRT_ITER ( 2);
-    WEBRTC_SPL_SQRT_ITER ( 1);
-    WEBRTC_SPL_SQRT_ITER ( 0);
-
-    return root >> 1;
-}

src/common_audio/signal_processing_library/main/source/spl_version.c

@@ -1,25 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_get_version().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include <string.h>
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_get_version(char* version, WebRtc_Word16 length_in_bytes)
-{
-    strncpy(version, "1.2.0", length_in_bytes);
-    return 0;
-}

src/common_audio/signal_processing_library/main/source/vector_scaling_operations.c

@@ -1,151 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the functions
- * WebRtcSpl_VectorBitShiftW16()
- * WebRtcSpl_VectorBitShiftW32()
- * WebRtcSpl_VectorBitShiftW32ToW16()
- * WebRtcSpl_ScaleVector()
- * WebRtcSpl_ScaleVectorWithSat()
- * WebRtcSpl_ScaleAndAddVectors()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_VectorBitShiftW16(WebRtc_Word16 *res,
-                             WebRtc_Word16 length,
-                             G_CONST WebRtc_Word16 *in,
-                             WebRtc_Word16 right_shifts)
-{
-    int i;
-
-    if (right_shifts > 0)
-    {
-        for (i = length; i > 0; i--)
-        {
-            (*res++) = ((*in++) >> right_shifts);
-        }
-    } else
-    {
-        for (i = length; i > 0; i--)
-        {
-            (*res++) = ((*in++) << (-right_shifts));
-        }
-    }
-}
-
-void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32 *out_vector,
-                             WebRtc_Word16 vector_length,
-                             G_CONST WebRtc_Word32 *in_vector,
-                             WebRtc_Word16 right_shifts)
-{
-    int i;
-
-    if (right_shifts > 0)
-    {
-        for (i = vector_length; i > 0; i--)
-        {
-            (*out_vector++) = ((*in_vector++) >> right_shifts);
-        }
-    } else
-    {
-        for (i = vector_length; i > 0; i--)
-        {
-            (*out_vector++) = ((*in_vector++) << (-right_shifts));
-        }
-    }
-}
-
-void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16 *res,
-                                  WebRtc_Word16 length,
-                                  G_CONST WebRtc_Word32 *in,
-                                  WebRtc_Word16 right_shifts)
-{
-    int i;
-
-    if (right_shifts >= 0)
-    {
-        for (i = length; i > 0; i--)
-        {
-            (*res++) = (WebRtc_Word16)((*in++) >> right_shifts);
-        }
-    } else
-    {
-        WebRtc_Word16 left_shifts = -right_shifts;
-        for (i = length; i > 0; i--)
-        {
-            (*res++) = (WebRtc_Word16)((*in++) << left_shifts);
-        }
-    }
-}
-
-void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector,
-                           WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
-                           WebRtc_Word16 right_shifts)
-{
-    // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
-    int i;
-    G_CONST WebRtc_Word16 *inptr;
-    WebRtc_Word16 *outptr;
-
-    inptr = in_vector;
-    outptr = out_vector;
-
-    for (i = 0; i < in_vector_length; i++)
-    {
-        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
-    }
-}
-
-void WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector,
-                                 WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
-                                 WebRtc_Word16 right_shifts)
-{
-    // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
-    int i;
-    WebRtc_Word32 tmpW32;
-    G_CONST WebRtc_Word16 *inptr;
-    WebRtc_Word16 *outptr;
-
-    inptr = in_vector;
-    outptr = out_vector;
-
-    for (i = 0; i < in_vector_length; i++)
-    {
-        tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
-        (*outptr++) = WebRtcSpl_SatW32ToW16(tmpW32);
-    }
-}
-
-void WebRtcSpl_ScaleAndAddVectors(G_CONST WebRtc_Word16 *in1, WebRtc_Word16 gain1, int shift1,
-                                  G_CONST WebRtc_Word16 *in2, WebRtc_Word16 gain2, int shift2,
-                                  WebRtc_Word16 *out, int vector_length)
-{
-    // Performs vector operation: out = (gain1*in1)>>shift1 + (gain2*in2)>>shift2
-    int i;
-    G_CONST WebRtc_Word16 *in1ptr;
-    G_CONST WebRtc_Word16 *in2ptr;
-    WebRtc_Word16 *outptr;
-
-    in1ptr = in1;
-    in2ptr = in2;
-    outptr = out;
-
-    for (i = 0; i < vector_length; i++)
-    {
-        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gain1, *in1ptr++, shift1)
-                + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gain2, *in2ptr++, shift2);
-    }
-}

src/common_audio/signal_processing_library/main/source/webrtc_fft_t_1024_8.c

@@ -1,704 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the Q14 radix-8 tables used in ARM9e optimizations.
- *
- */
-
-extern const int s_Q14S_8;
-const int s_Q14S_8 = 1024;
-extern const unsigned short t_Q14S_8[2032];
-const unsigned short t_Q14S_8[2032] = {
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
-  0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
-  0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
-  0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
-  0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
-  0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
-  0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x396b,0x0646 ,0x3cc8,0x0324 ,0x35eb,0x0964 ,
-  0x3249,0x0c7c ,0x396b,0x0646 ,0x2aaa,0x1294 ,
-  0x2aaa,0x1294 ,0x35eb,0x0964 ,0x1e7e,0x1b5d ,
-  0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
-  0x1a46,0x1e2b ,0x2e88,0x0f8d ,0x0471,0x2afb ,
-  0x11a8,0x238e ,0x2aaa,0x1294 ,0xf721,0x3179 ,
-  0x08df,0x289a ,0x26b3,0x1590 ,0xea02,0x36e5 ,
-  0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
-  0xf721,0x3179 ,0x1e7e,0x1b5d ,0xd178,0x3e15 ,
-  0xee58,0x3537 ,0x1a46,0x1e2b ,0xc695,0x3fb1 ,
-  0xe5ba,0x3871 ,0x15fe,0x20e7 ,0xbcf0,0x3fec ,
-  0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
-  0xd556,0x3d3f ,0x0d48,0x2620 ,0xae2e,0x3c42 ,
-  0xcdb7,0x3ec5 ,0x08df,0x289a ,0xa963,0x3871 ,
-  0xc695,0x3fb1 ,0x0471,0x2afb ,0xa678,0x3368 ,
-  0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
-  0xba09,0x3fb1 ,0xfb8f,0x2f6c ,0xa678,0x2620 ,
-  0xb4be,0x3ec5 ,0xf721,0x3179 ,0xa963,0x1e2b ,
-  0xb02d,0x3d3f ,0xf2b8,0x3368 ,0xae2e,0x1590 ,
-  0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
-  0xa963,0x3871 ,0xea02,0x36e5 ,0xbcf0,0x0324 ,
-  0xa73b,0x3537 ,0xe5ba,0x3871 ,0xc695,0xf9ba ,
-  0xa5ed,0x3179 ,0xe182,0x39db ,0xd178,0xf073 ,
-  0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
-  0xa5ed,0x289a ,0xd94d,0x3c42 ,0xea02,0xdf19 ,
-  0xa73b,0x238e ,0xd556,0x3d3f ,0xf721,0xd766 ,
-  0xa963,0x1e2b ,0xd178,0x3e15 ,0x0471,0xd094 ,
-  0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
-  0xb02d,0x1294 ,0xca15,0x3f4f ,0x1e7e,0xc625 ,
-  0xb4be,0x0c7c ,0xc695,0x3fb1 ,0x2aaa,0xc2c1 ,
-  0xba09,0x0646 ,0xc338,0x3fec ,0x35eb,0xc0b1 ,
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x3e69,0x0192 ,0x3f36,0x00c9 ,0x3d9a,0x025b ,
-  0x3cc8,0x0324 ,0x3e69,0x0192 ,0x3b1e,0x04b5 ,
-  0x3b1e,0x04b5 ,0x3d9a,0x025b ,0x388e,0x070e ,
-  0x396b,0x0646 ,0x3cc8,0x0324 ,0x35eb,0x0964 ,
-  0x37af,0x07d6 ,0x3bf4,0x03ed ,0x3334,0x0bb7 ,
-  0x35eb,0x0964 ,0x3b1e,0x04b5 ,0x306c,0x0e06 ,
-  0x341e,0x0af1 ,0x3a46,0x057e ,0x2d93,0x1050 ,
-  0x3249,0x0c7c ,0x396b,0x0646 ,0x2aaa,0x1294 ,
-  0x306c,0x0e06 ,0x388e,0x070e ,0x27b3,0x14d2 ,
-  0x2e88,0x0f8d ,0x37af,0x07d6 ,0x24ae,0x1709 ,
-  0x2c9d,0x1112 ,0x36ce,0x089d ,0x219c,0x1937 ,
-  0x2aaa,0x1294 ,0x35eb,0x0964 ,0x1e7e,0x1b5d ,
-  0x28b2,0x1413 ,0x3505,0x0a2b ,0x1b56,0x1d79 ,
-  0x26b3,0x1590 ,0x341e,0x0af1 ,0x1824,0x1f8c ,
-  0x24ae,0x1709 ,0x3334,0x0bb7 ,0x14ea,0x2193 ,
-  0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
-  0x2093,0x19ef ,0x315b,0x0d41 ,0x0e61,0x257e ,
-  0x1e7e,0x1b5d ,0x306c,0x0e06 ,0x0b14,0x2760 ,
-  0x1c64,0x1cc6 ,0x2f7b,0x0eca ,0x07c4,0x2935 ,
-  0x1a46,0x1e2b ,0x2e88,0x0f8d ,0x0471,0x2afb ,
-  0x1824,0x1f8c ,0x2d93,0x1050 ,0x011c,0x2cb2 ,
-  0x15fe,0x20e7 ,0x2c9d,0x1112 ,0xfdc7,0x2e5a ,
-  0x13d5,0x223d ,0x2ba4,0x11d3 ,0xfa73,0x2ff2 ,
-  0x11a8,0x238e ,0x2aaa,0x1294 ,0xf721,0x3179 ,
-  0x0f79,0x24da ,0x29af,0x1354 ,0xf3d2,0x32ef ,
-  0x0d48,0x2620 ,0x28b2,0x1413 ,0xf087,0x3453 ,
-  0x0b14,0x2760 ,0x27b3,0x14d2 ,0xed41,0x35a5 ,
-  0x08df,0x289a ,0x26b3,0x1590 ,0xea02,0x36e5 ,
-  0x06a9,0x29ce ,0x25b1,0x164c ,0xe6cb,0x3812 ,
-  0x0471,0x2afb ,0x24ae,0x1709 ,0xe39c,0x392b ,
-  0x0239,0x2c21 ,0x23a9,0x17c4 ,0xe077,0x3a30 ,
-  0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
-  0xfdc7,0x2e5a ,0x219c,0x1937 ,0xda4f,0x3bfd ,
-  0xfb8f,0x2f6c ,0x2093,0x19ef ,0xd74e,0x3cc5 ,
-  0xf957,0x3076 ,0x1f89,0x1aa7 ,0xd45c,0x3d78 ,
-  0xf721,0x3179 ,0x1e7e,0x1b5d ,0xd178,0x3e15 ,
-  0xf4ec,0x3274 ,0x1d72,0x1c12 ,0xcea5,0x3e9d ,
-  0xf2b8,0x3368 ,0x1c64,0x1cc6 ,0xcbe2,0x3f0f ,
-  0xf087,0x3453 ,0x1b56,0x1d79 ,0xc932,0x3f6b ,
-  0xee58,0x3537 ,0x1a46,0x1e2b ,0xc695,0x3fb1 ,
-  0xec2b,0x3612 ,0x1935,0x1edc ,0xc40c,0x3fe1 ,
-  0xea02,0x36e5 ,0x1824,0x1f8c ,0xc197,0x3ffb ,
-  0xe7dc,0x37b0 ,0x1711,0x203a ,0xbf38,0x3fff ,
-  0xe5ba,0x3871 ,0x15fe,0x20e7 ,0xbcf0,0x3fec ,
-  0xe39c,0x392b ,0x14ea,0x2193 ,0xbabf,0x3fc4 ,
-  0xe182,0x39db ,0x13d5,0x223d ,0xb8a6,0x3f85 ,
-  0xdf6d,0x3a82 ,0x12bf,0x22e7 ,0xb6a5,0x3f30 ,
-  0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
-  0xdb52,0x3bb6 ,0x1091,0x2435 ,0xb2f2,0x3e45 ,
-  0xd94d,0x3c42 ,0x0f79,0x24da ,0xb140,0x3daf ,
-  0xd74e,0x3cc5 ,0x0e61,0x257e ,0xafa9,0x3d03 ,
-  0xd556,0x3d3f ,0x0d48,0x2620 ,0xae2e,0x3c42 ,
-  0xd363,0x3daf ,0x0c2e,0x26c1 ,0xacd0,0x3b6d ,
-  0xd178,0x3e15 ,0x0b14,0x2760 ,0xab8e,0x3a82 ,
-  0xcf94,0x3e72 ,0x09fa,0x27fe ,0xaa6a,0x3984 ,
-  0xcdb7,0x3ec5 ,0x08df,0x289a ,0xa963,0x3871 ,
-  0xcbe2,0x3f0f ,0x07c4,0x2935 ,0xa87b,0x374b ,
-  0xca15,0x3f4f ,0x06a9,0x29ce ,0xa7b1,0x3612 ,
-  0xc851,0x3f85 ,0x058d,0x2a65 ,0xa705,0x34c6 ,
-  0xc695,0x3fb1 ,0x0471,0x2afb ,0xa678,0x3368 ,
-  0xc4e2,0x3fd4 ,0x0355,0x2b8f ,0xa60b,0x31f8 ,
-  0xc338,0x3fec ,0x0239,0x2c21 ,0xa5bc,0x3076 ,
-  0xc197,0x3ffb ,0x011c,0x2cb2 ,0xa58d,0x2ee4 ,
-  0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
-  0xbe73,0x3ffb ,0xfee4,0x2dcf ,0xa58d,0x2b8f ,
-  0xbcf0,0x3fec ,0xfdc7,0x2e5a ,0xa5bc,0x29ce ,
-  0xbb77,0x3fd4 ,0xfcab,0x2ee4 ,0xa60b,0x27fe ,
-  0xba09,0x3fb1 ,0xfb8f,0x2f6c ,0xa678,0x2620 ,
-  0xb8a6,0x3f85 ,0xfa73,0x2ff2 ,0xa705,0x2435 ,
-  0xb74d,0x3f4f ,0xf957,0x3076 ,0xa7b1,0x223d ,
-  0xb600,0x3f0f ,0xf83c,0x30f9 ,0xa87b,0x203a ,
-  0xb4be,0x3ec5 ,0xf721,0x3179 ,0xa963,0x1e2b ,
-  0xb388,0x3e72 ,0xf606,0x31f8 ,0xaa6a,0x1c12 ,
-  0xb25e,0x3e15 ,0xf4ec,0x3274 ,0xab8e,0x19ef ,
-  0xb140,0x3daf ,0xf3d2,0x32ef ,0xacd0,0x17c4 ,
-  0xb02d,0x3d3f ,0xf2b8,0x3368 ,0xae2e,0x1590 ,
-  0xaf28,0x3cc5 ,0xf19f,0x33df ,0xafa9,0x1354 ,
-  0xae2e,0x3c42 ,0xf087,0x3453 ,0xb140,0x1112 ,
-  0xad41,0x3bb6 ,0xef6f,0x34c6 ,0xb2f2,0x0eca ,
-  0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
-  0xab8e,0x3a82 ,0xed41,0x35a5 ,0xb6a5,0x0a2b ,
-  0xaac8,0x39db ,0xec2b,0x3612 ,0xb8a6,0x07d6 ,
-  0xaa0f,0x392b ,0xeb16,0x367d ,0xbabf,0x057e ,
-  0xa963,0x3871 ,0xea02,0x36e5 ,0xbcf0,0x0324 ,
-  0xa8c5,0x37b0 ,0xe8ef,0x374b ,0xbf38,0x00c9 ,
-  0xa834,0x36e5 ,0xe7dc,0x37b0 ,0xc197,0xfe6e ,
-  0xa7b1,0x3612 ,0xe6cb,0x3812 ,0xc40c,0xfc13 ,
-  0xa73b,0x3537 ,0xe5ba,0x3871 ,0xc695,0xf9ba ,
-  0xa6d3,0x3453 ,0xe4aa,0x38cf ,0xc932,0xf763 ,
-  0xa678,0x3368 ,0xe39c,0x392b ,0xcbe2,0xf50f ,
-  0xa62c,0x3274 ,0xe28e,0x3984 ,0xcea5,0xf2bf ,
-  0xa5ed,0x3179 ,0xe182,0x39db ,0xd178,0xf073 ,
-  0xa5bc,0x3076 ,0xe077,0x3a30 ,0xd45c,0xee2d ,
-  0xa599,0x2f6c ,0xdf6d,0x3a82 ,0xd74e,0xebed ,
-  0xa585,0x2e5a ,0xde64,0x3ad3 ,0xda4f,0xe9b4 ,
-  0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
-  0xa585,0x2c21 ,0xdc57,0x3b6d ,0xe077,0xe559 ,
-  0xa599,0x2afb ,0xdb52,0x3bb6 ,0xe39c,0xe33a ,
-  0xa5bc,0x29ce ,0xda4f,0x3bfd ,0xe6cb,0xe124 ,
-  0xa5ed,0x289a ,0xd94d,0x3c42 ,0xea02,0xdf19 ,
-  0xa62c,0x2760 ,0xd84d,0x3c85 ,0xed41,0xdd19 ,
-  0xa678,0x2620 ,0xd74e,0x3cc5 ,0xf087,0xdb26 ,
-  0xa6d3,0x24da ,0xd651,0x3d03 ,0xf3d2,0xd93f ,
-  0xa73b,0x238e ,0xd556,0x3d3f ,0xf721,0xd766 ,
-  0xa7b1,0x223d ,0xd45c,0x3d78 ,0xfa73,0xd59b ,
-  0xa834,0x20e7 ,0xd363,0x3daf ,0xfdc7,0xd3df ,
-  0xa8c5,0x1f8c ,0xd26d,0x3de3 ,0x011c,0xd231 ,
-  0xa963,0x1e2b ,0xd178,0x3e15 ,0x0471,0xd094 ,
-  0xaa0f,0x1cc6 ,0xd085,0x3e45 ,0x07c4,0xcf07 ,
-  0xaac8,0x1b5d ,0xcf94,0x3e72 ,0x0b14,0xcd8c ,
-  0xab8e,0x19ef ,0xcea5,0x3e9d ,0x0e61,0xcc21 ,
-  0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
-  0xad41,0x1709 ,0xcccc,0x3eeb ,0x14ea,0xc983 ,
-  0xae2e,0x1590 ,0xcbe2,0x3f0f ,0x1824,0xc850 ,
-  0xaf28,0x1413 ,0xcafb,0x3f30 ,0x1b56,0xc731 ,
-  0xb02d,0x1294 ,0xca15,0x3f4f ,0x1e7e,0xc625 ,
-  0xb140,0x1112 ,0xc932,0x3f6b ,0x219c,0xc52d ,
-  0xb25e,0x0f8d ,0xc851,0x3f85 ,0x24ae,0xc44a ,
-  0xb388,0x0e06 ,0xc772,0x3f9c ,0x27b3,0xc37b ,
-  0xb4be,0x0c7c ,0xc695,0x3fb1 ,0x2aaa,0xc2c1 ,
-  0xb600,0x0af1 ,0xc5ba,0x3fc4 ,0x2d93,0xc21d ,
-  0xb74d,0x0964 ,0xc4e2,0x3fd4 ,0x306c,0xc18e ,
-  0xb8a6,0x07d6 ,0xc40c,0x3fe1 ,0x3334,0xc115 ,
-  0xba09,0x0646 ,0xc338,0x3fec ,0x35eb,0xc0b1 ,
-  0xbb77,0x04b5 ,0xc266,0x3ff5 ,0x388e,0xc064 ,
-  0xbcf0,0x0324 ,0xc197,0x3ffb ,0x3b1e,0xc02c ,
-  0xbe73,0x0192 ,0xc0ca,0x3fff ,0x3d9a,0xc00b ,
-  0x4000,0x0000 ,0x3f9b,0x0065 ,0x3f36,0x00c9 ,
-  0x3ed0,0x012e ,0x3e69,0x0192 ,0x3e02,0x01f7 ,
-  0x3d9a,0x025b ,0x3d31,0x02c0 ,0x3cc8,0x0324 ,
-  0x3c5f,0x0388 ,0x3bf4,0x03ed ,0x3b8a,0x0451 ,
-  0x3b1e,0x04b5 ,0x3ab2,0x051a ,0x3a46,0x057e ,
-  0x39d9,0x05e2 ,0x396b,0x0646 ,0x38fd,0x06aa ,
-  0x388e,0x070e ,0x381f,0x0772 ,0x37af,0x07d6 ,
-  0x373f,0x0839 ,0x36ce,0x089d ,0x365d,0x0901 ,
-  0x35eb,0x0964 ,0x3578,0x09c7 ,0x3505,0x0a2b ,
-  0x3492,0x0a8e ,0x341e,0x0af1 ,0x33a9,0x0b54 ,
-  0x3334,0x0bb7 ,0x32bf,0x0c1a ,0x3249,0x0c7c ,
-  0x31d2,0x0cdf ,0x315b,0x0d41 ,0x30e4,0x0da4 ,
-  0x306c,0x0e06 ,0x2ff4,0x0e68 ,0x2f7b,0x0eca ,
-  0x2f02,0x0f2b ,0x2e88,0x0f8d ,0x2e0e,0x0fee ,
-  0x2d93,0x1050 ,0x2d18,0x10b1 ,0x2c9d,0x1112 ,
-  0x2c21,0x1173 ,0x2ba4,0x11d3 ,0x2b28,0x1234 ,
-  0x2aaa,0x1294 ,0x2a2d,0x12f4 ,0x29af,0x1354 ,
-  0x2931,0x13b4 ,0x28b2,0x1413 ,0x2833,0x1473 ,
-  0x27b3,0x14d2 ,0x2733,0x1531 ,0x26b3,0x1590 ,
-  0x2632,0x15ee ,0x25b1,0x164c ,0x252f,0x16ab ,
-  0x24ae,0x1709 ,0x242b,0x1766 ,0x23a9,0x17c4 ,
-  0x2326,0x1821 ,0x22a3,0x187e ,0x221f,0x18db ,
-  0x219c,0x1937 ,0x2117,0x1993 ,0x2093,0x19ef ,
-  0x200e,0x1a4b ,0x1f89,0x1aa7 ,0x1f04,0x1b02 ,
-  0x1e7e,0x1b5d ,0x1df8,0x1bb8 ,0x1d72,0x1c12 ,
-  0x1ceb,0x1c6c ,0x1c64,0x1cc6 ,0x1bdd,0x1d20 ,
-  0x1b56,0x1d79 ,0x1ace,0x1dd3 ,0x1a46,0x1e2b ,
-  0x19be,0x1e84 ,0x1935,0x1edc ,0x18ad,0x1f34 ,
-  0x1824,0x1f8c ,0x179b,0x1fe3 ,0x1711,0x203a ,
-  0x1688,0x2091 ,0x15fe,0x20e7 ,0x1574,0x213d ,
-  0x14ea,0x2193 ,0x145f,0x21e8 ,0x13d5,0x223d ,
-  0x134a,0x2292 ,0x12bf,0x22e7 ,0x1234,0x233b ,
-  0x11a8,0x238e ,0x111d,0x23e2 ,0x1091,0x2435 ,
-  0x1005,0x2488 ,0x0f79,0x24da ,0x0eed,0x252c ,
-  0x0e61,0x257e ,0x0dd4,0x25cf ,0x0d48,0x2620 ,
-  0x0cbb,0x2671 ,0x0c2e,0x26c1 ,0x0ba1,0x2711 ,
-  0x0b14,0x2760 ,0x0a87,0x27af ,0x09fa,0x27fe ,
-  0x096d,0x284c ,0x08df,0x289a ,0x0852,0x28e7 ,
-  0x07c4,0x2935 ,0x0736,0x2981 ,0x06a9,0x29ce ,
-  0x061b,0x2a1a ,0x058d,0x2a65 ,0x04ff,0x2ab0 ,
-  0x0471,0x2afb ,0x03e3,0x2b45 ,0x0355,0x2b8f ,
-  0x02c7,0x2bd8 ,0x0239,0x2c21 ,0x01aa,0x2c6a ,
-  0x011c,0x2cb2 ,0x008e,0x2cfa ,0x0000,0x2d41 ,
-  0xff72,0x2d88 ,0xfee4,0x2dcf ,0xfe56,0x2e15 ,
-  0xfdc7,0x2e5a ,0xfd39,0x2e9f ,0xfcab,0x2ee4 ,
-  0xfc1d,0x2f28 ,0xfb8f,0x2f6c ,0xfb01,0x2faf ,
-  0xfa73,0x2ff2 ,0xf9e5,0x3034 ,0xf957,0x3076 ,
-  0xf8ca,0x30b8 ,0xf83c,0x30f9 ,0xf7ae,0x3139 ,
-  0xf721,0x3179 ,0xf693,0x31b9 ,0xf606,0x31f8 ,
-  0xf579,0x3236 ,0xf4ec,0x3274 ,0xf45f,0x32b2 ,
-  0xf3d2,0x32ef ,0xf345,0x332c ,0xf2b8,0x3368 ,
-  0xf22c,0x33a3 ,0xf19f,0x33df ,0xf113,0x3419 ,
-  0xf087,0x3453 ,0xeffb,0x348d ,0xef6f,0x34c6 ,
-  0xeee3,0x34ff ,0xee58,0x3537 ,0xedcc,0x356e ,
-  0xed41,0x35a5 ,0xecb6,0x35dc ,0xec2b,0x3612 ,
-  0xeba1,0x3648 ,0xeb16,0x367d ,0xea8c,0x36b1 ,
-  0xea02,0x36e5 ,0xe978,0x3718 ,0xe8ef,0x374b ,
-  0xe865,0x377e ,0xe7dc,0x37b0 ,0xe753,0x37e1 ,
-  0xe6cb,0x3812 ,0xe642,0x3842 ,0xe5ba,0x3871 ,
-  0xe532,0x38a1 ,0xe4aa,0x38cf ,0xe423,0x38fd ,
-  0xe39c,0x392b ,0xe315,0x3958 ,0xe28e,0x3984 ,
-  0xe208,0x39b0 ,0xe182,0x39db ,0xe0fc,0x3a06 ,
-  0xe077,0x3a30 ,0xdff2,0x3a59 ,0xdf6d,0x3a82 ,
-  0xdee9,0x3aab ,0xde64,0x3ad3 ,0xdde1,0x3afa ,
-  0xdd5d,0x3b21 ,0xdcda,0x3b47 ,0xdc57,0x3b6d ,
-  0xdbd5,0x3b92 ,0xdb52,0x3bb6 ,0xdad1,0x3bda ,
-  0xda4f,0x3bfd ,0xd9ce,0x3c20 ,0xd94d,0x3c42 ,
-  0xd8cd,0x3c64 ,0xd84d,0x3c85 ,0xd7cd,0x3ca5 ,
-  0xd74e,0x3cc5 ,0xd6cf,0x3ce4 ,0xd651,0x3d03 ,
-  0xd5d3,0x3d21 ,0xd556,0x3d3f ,0xd4d8,0x3d5b ,
-  0xd45c,0x3d78 ,0xd3df,0x3d93 ,0xd363,0x3daf ,
-  0xd2e8,0x3dc9 ,0xd26d,0x3de3 ,0xd1f2,0x3dfc ,
-  0xd178,0x3e15 ,0xd0fe,0x3e2d ,0xd085,0x3e45 ,
-  0xd00c,0x3e5c ,0xcf94,0x3e72 ,0xcf1c,0x3e88 ,
-  0xcea5,0x3e9d ,0xce2e,0x3eb1 ,0xcdb7,0x3ec5 ,
-  0xcd41,0x3ed8 ,0xcccc,0x3eeb ,0xcc57,0x3efd ,
-  0xcbe2,0x3f0f ,0xcb6e,0x3f20 ,0xcafb,0x3f30 ,
-  0xca88,0x3f40 ,0xca15,0x3f4f ,0xc9a3,0x3f5d ,
-  0xc932,0x3f6b ,0xc8c1,0x3f78 ,0xc851,0x3f85 ,
-  0xc7e1,0x3f91 ,0xc772,0x3f9c ,0xc703,0x3fa7 ,
-  0xc695,0x3fb1 ,0xc627,0x3fbb ,0xc5ba,0x3fc4 ,
-  0xc54e,0x3fcc ,0xc4e2,0x3fd4 ,0xc476,0x3fdb ,
-  0xc40c,0x3fe1 ,0xc3a1,0x3fe7 ,0xc338,0x3fec ,
-  0xc2cf,0x3ff1 ,0xc266,0x3ff5 ,0xc1fe,0x3ff8 ,
-  0xc197,0x3ffb ,0xc130,0x3ffd ,0xc0ca,0x3fff ,
-  0xc065,0x4000 ,0xc000,0x4000 ,0xbf9c,0x4000 ,
-  0xbf38,0x3fff ,0xbed5,0x3ffd ,0xbe73,0x3ffb ,
-  0xbe11,0x3ff8 ,0xbdb0,0x3ff5 ,0xbd50,0x3ff1 ,
-  0xbcf0,0x3fec ,0xbc91,0x3fe7 ,0xbc32,0x3fe1 ,
-  0xbbd4,0x3fdb ,0xbb77,0x3fd4 ,0xbb1b,0x3fcc ,
-  0xbabf,0x3fc4 ,0xba64,0x3fbb ,0xba09,0x3fb1 ,
-  0xb9af,0x3fa7 ,0xb956,0x3f9c ,0xb8fd,0x3f91 ,
-  0xb8a6,0x3f85 ,0xb84f,0x3f78 ,0xb7f8,0x3f6b ,
-  0xb7a2,0x3f5d ,0xb74d,0x3f4f ,0xb6f9,0x3f40 ,
-  0xb6a5,0x3f30 ,0xb652,0x3f20 ,0xb600,0x3f0f ,
-  0xb5af,0x3efd ,0xb55e,0x3eeb ,0xb50e,0x3ed8 ,
-  0xb4be,0x3ec5 ,0xb470,0x3eb1 ,0xb422,0x3e9d ,
-  0xb3d5,0x3e88 ,0xb388,0x3e72 ,0xb33d,0x3e5c ,
-  0xb2f2,0x3e45 ,0xb2a7,0x3e2d ,0xb25e,0x3e15 ,
-  0xb215,0x3dfc ,0xb1cd,0x3de3 ,0xb186,0x3dc9 ,
-  0xb140,0x3daf ,0xb0fa,0x3d93 ,0xb0b5,0x3d78 ,
-  0xb071,0x3d5b ,0xb02d,0x3d3f ,0xafeb,0x3d21 ,
-  0xafa9,0x3d03 ,0xaf68,0x3ce4 ,0xaf28,0x3cc5 ,
-  0xaee8,0x3ca5 ,0xaea9,0x3c85 ,0xae6b,0x3c64 ,
-  0xae2e,0x3c42 ,0xadf2,0x3c20 ,0xadb6,0x3bfd ,
-  0xad7b,0x3bda ,0xad41,0x3bb6 ,0xad08,0x3b92 ,
-  0xacd0,0x3b6d ,0xac98,0x3b47 ,0xac61,0x3b21 ,
-  0xac2b,0x3afa ,0xabf6,0x3ad3 ,0xabc2,0x3aab ,
-  0xab8e,0x3a82 ,0xab5b,0x3a59 ,0xab29,0x3a30 ,
-  0xaaf8,0x3a06 ,0xaac8,0x39db ,0xaa98,0x39b0 ,
-  0xaa6a,0x3984 ,0xaa3c,0x3958 ,0xaa0f,0x392b ,
-  0xa9e3,0x38fd ,0xa9b7,0x38cf ,0xa98d,0x38a1 ,
-  0xa963,0x3871 ,0xa93a,0x3842 ,0xa912,0x3812 ,
-  0xa8eb,0x37e1 ,0xa8c5,0x37b0 ,0xa89f,0x377e ,
-  0xa87b,0x374b ,0xa857,0x3718 ,0xa834,0x36e5 ,
-  0xa812,0x36b1 ,0xa7f1,0x367d ,0xa7d0,0x3648 ,
-  0xa7b1,0x3612 ,0xa792,0x35dc ,0xa774,0x35a5 ,
-  0xa757,0x356e ,0xa73b,0x3537 ,0xa71f,0x34ff ,
-  0xa705,0x34c6 ,0xa6eb,0x348d ,0xa6d3,0x3453 ,
-  0xa6bb,0x3419 ,0xa6a4,0x33df ,0xa68e,0x33a3 ,
-  0xa678,0x3368 ,0xa664,0x332c ,0xa650,0x32ef ,
-  0xa63e,0x32b2 ,0xa62c,0x3274 ,0xa61b,0x3236 ,
-  0xa60b,0x31f8 ,0xa5fb,0x31b9 ,0xa5ed,0x3179 ,
-  0xa5e0,0x3139 ,0xa5d3,0x30f9 ,0xa5c7,0x30b8 ,
-  0xa5bc,0x3076 ,0xa5b2,0x3034 ,0xa5a9,0x2ff2 ,
-  0xa5a1,0x2faf ,0xa599,0x2f6c ,0xa593,0x2f28 ,
-  0xa58d,0x2ee4 ,0xa588,0x2e9f ,0xa585,0x2e5a ,
-  0xa581,0x2e15 ,0xa57f,0x2dcf ,0xa57e,0x2d88 ,
-  0xa57e,0x2d41 ,0xa57e,0x2cfa ,0xa57f,0x2cb2 ,
-  0xa581,0x2c6a ,0xa585,0x2c21 ,0xa588,0x2bd8 ,
-  0xa58d,0x2b8f ,0xa593,0x2b45 ,0xa599,0x2afb ,
-  0xa5a1,0x2ab0 ,0xa5a9,0x2a65 ,0xa5b2,0x2a1a ,
-  0xa5bc,0x29ce ,0xa5c7,0x2981 ,0xa5d3,0x2935 ,
-  0xa5e0,0x28e7 ,0xa5ed,0x289a ,0xa5fb,0x284c ,
-  0xa60b,0x27fe ,0xa61b,0x27af ,0xa62c,0x2760 ,
-  0xa63e,0x2711 ,0xa650,0x26c1 ,0xa664,0x2671 ,
-  0xa678,0x2620 ,0xa68e,0x25cf ,0xa6a4,0x257e ,
-  0xa6bb,0x252c ,0xa6d3,0x24da ,0xa6eb,0x2488 ,
-  0xa705,0x2435 ,0xa71f,0x23e2 ,0xa73b,0x238e ,
-  0xa757,0x233b ,0xa774,0x22e7 ,0xa792,0x2292 ,
-  0xa7b1,0x223d ,0xa7d0,0x21e8 ,0xa7f1,0x2193 ,
-  0xa812,0x213d ,0xa834,0x20e7 ,0xa857,0x2091 ,
-  0xa87b,0x203a ,0xa89f,0x1fe3 ,0xa8c5,0x1f8c ,
-  0xa8eb,0x1f34 ,0xa912,0x1edc ,0xa93a,0x1e84 ,
-  0xa963,0x1e2b ,0xa98d,0x1dd3 ,0xa9b7,0x1d79 ,
-  0xa9e3,0x1d20 ,0xaa0f,0x1cc6 ,0xaa3c,0x1c6c ,
-  0xaa6a,0x1c12 ,0xaa98,0x1bb8 ,0xaac8,0x1b5d ,
-  0xaaf8,0x1b02 ,0xab29,0x1aa7 ,0xab5b,0x1a4b ,
-  0xab8e,0x19ef ,0xabc2,0x1993 ,0xabf6,0x1937 ,
-  0xac2b,0x18db ,0xac61,0x187e ,0xac98,0x1821 ,
-  0xacd0,0x17c4 ,0xad08,0x1766 ,0xad41,0x1709 ,
-  0xad7b,0x16ab ,0xadb6,0x164c ,0xadf2,0x15ee ,
-  0xae2e,0x1590 ,0xae6b,0x1531 ,0xaea9,0x14d2 ,
-  0xaee8,0x1473 ,0xaf28,0x1413 ,0xaf68,0x13b4 ,
-  0xafa9,0x1354 ,0xafeb,0x12f4 ,0xb02d,0x1294 ,
-  0xb071,0x1234 ,0xb0b5,0x11d3 ,0xb0fa,0x1173 ,
-  0xb140,0x1112 ,0xb186,0x10b1 ,0xb1cd,0x1050 ,
-  0xb215,0x0fee ,0xb25e,0x0f8d ,0xb2a7,0x0f2b ,
-  0xb2f2,0x0eca ,0xb33d,0x0e68 ,0xb388,0x0e06 ,
-  0xb3d5,0x0da4 ,0xb422,0x0d41 ,0xb470,0x0cdf ,
-  0xb4be,0x0c7c ,0xb50e,0x0c1a ,0xb55e,0x0bb7 ,
-  0xb5af,0x0b54 ,0xb600,0x0af1 ,0xb652,0x0a8e ,
-  0xb6a5,0x0a2b ,0xb6f9,0x09c7 ,0xb74d,0x0964 ,
-  0xb7a2,0x0901 ,0xb7f8,0x089d ,0xb84f,0x0839 ,
-  0xb8a6,0x07d6 ,0xb8fd,0x0772 ,0xb956,0x070e ,
-  0xb9af,0x06aa ,0xba09,0x0646 ,0xba64,0x05e2 ,
-  0xbabf,0x057e ,0xbb1b,0x051a ,0xbb77,0x04b5 ,
-  0xbbd4,0x0451 ,0xbc32,0x03ed ,0xbc91,0x0388 ,
-  0xbcf0,0x0324 ,0xbd50,0x02c0 ,0xbdb0,0x025b ,
-  0xbe11,0x01f7 ,0xbe73,0x0192 ,0xbed5,0x012e ,
-  0xbf38,0x00c9 ,0xbf9c,0x0065 };
-
-
-extern const int s_Q14R_8;
-const int s_Q14R_8 = 1024;
-extern const unsigned short t_Q14R_8[2032];
-const unsigned short t_Q14R_8[2032] = {
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
-  0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
-  0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
-  0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
-  0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
-  0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
-  0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x3fb1,0x0646 ,0x3fec,0x0324 ,0x3f4f,0x0964 ,
-  0x3ec5,0x0c7c ,0x3fb1,0x0646 ,0x3d3f,0x1294 ,
-  0x3d3f,0x1294 ,0x3f4f,0x0964 ,0x39db,0x1b5d ,
-  0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
-  0x3871,0x1e2b ,0x3e15,0x0f8d ,0x2f6c,0x2afb ,
-  0x3537,0x238e ,0x3d3f,0x1294 ,0x289a,0x3179 ,
-  0x3179,0x289a ,0x3c42,0x1590 ,0x20e7,0x36e5 ,
-  0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
-  0x289a,0x3179 ,0x39db,0x1b5d ,0x0f8d,0x3e15 ,
-  0x238e,0x3537 ,0x3871,0x1e2b ,0x0646,0x3fb1 ,
-  0x1e2b,0x3871 ,0x36e5,0x20e7 ,0xfcdc,0x3fec ,
-  0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
-  0x1294,0x3d3f ,0x3368,0x2620 ,0xea70,0x3c42 ,
-  0x0c7c,0x3ec5 ,0x3179,0x289a ,0xe1d5,0x3871 ,
-  0x0646,0x3fb1 ,0x2f6c,0x2afb ,0xd9e0,0x3368 ,
-  0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
-  0xf9ba,0x3fb1 ,0x2afb,0x2f6c ,0xcc98,0x2620 ,
-  0xf384,0x3ec5 ,0x289a,0x3179 ,0xc78f,0x1e2b ,
-  0xed6c,0x3d3f ,0x2620,0x3368 ,0xc3be,0x1590 ,
-  0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
-  0xe1d5,0x3871 ,0x20e7,0x36e5 ,0xc014,0x0324 ,
-  0xdc72,0x3537 ,0x1e2b,0x3871 ,0xc04f,0xf9ba ,
-  0xd766,0x3179 ,0x1b5d,0x39db ,0xc1eb,0xf073 ,
-  0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
-  0xce87,0x289a ,0x1590,0x3c42 ,0xc91b,0xdf19 ,
-  0xcac9,0x238e ,0x1294,0x3d3f ,0xce87,0xd766 ,
-  0xc78f,0x1e2b ,0x0f8d,0x3e15 ,0xd505,0xd094 ,
-  0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
-  0xc2c1,0x1294 ,0x0964,0x3f4f ,0xe4a3,0xc625 ,
-  0xc13b,0x0c7c ,0x0646,0x3fb1 ,0xed6c,0xc2c1 ,
-  0xc04f,0x0646 ,0x0324,0x3fec ,0xf69c,0xc0b1 ,
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x3ffb,0x0192 ,0x3fff,0x00c9 ,0x3ff5,0x025b ,
-  0x3fec,0x0324 ,0x3ffb,0x0192 ,0x3fd4,0x04b5 ,
-  0x3fd4,0x04b5 ,0x3ff5,0x025b ,0x3f9c,0x070e ,
-  0x3fb1,0x0646 ,0x3fec,0x0324 ,0x3f4f,0x0964 ,
-  0x3f85,0x07d6 ,0x3fe1,0x03ed ,0x3eeb,0x0bb7 ,
-  0x3f4f,0x0964 ,0x3fd4,0x04b5 ,0x3e72,0x0e06 ,
-  0x3f0f,0x0af1 ,0x3fc4,0x057e ,0x3de3,0x1050 ,
-  0x3ec5,0x0c7c ,0x3fb1,0x0646 ,0x3d3f,0x1294 ,
-  0x3e72,0x0e06 ,0x3f9c,0x070e ,0x3c85,0x14d2 ,
-  0x3e15,0x0f8d ,0x3f85,0x07d6 ,0x3bb6,0x1709 ,
-  0x3daf,0x1112 ,0x3f6b,0x089d ,0x3ad3,0x1937 ,
-  0x3d3f,0x1294 ,0x3f4f,0x0964 ,0x39db,0x1b5d ,
-  0x3cc5,0x1413 ,0x3f30,0x0a2b ,0x38cf,0x1d79 ,
-  0x3c42,0x1590 ,0x3f0f,0x0af1 ,0x37b0,0x1f8c ,
-  0x3bb6,0x1709 ,0x3eeb,0x0bb7 ,0x367d,0x2193 ,
-  0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
-  0x3a82,0x19ef ,0x3e9d,0x0d41 ,0x33df,0x257e ,
-  0x39db,0x1b5d ,0x3e72,0x0e06 ,0x3274,0x2760 ,
-  0x392b,0x1cc6 ,0x3e45,0x0eca ,0x30f9,0x2935 ,
-  0x3871,0x1e2b ,0x3e15,0x0f8d ,0x2f6c,0x2afb ,
-  0x37b0,0x1f8c ,0x3de3,0x1050 ,0x2dcf,0x2cb2 ,
-  0x36e5,0x20e7 ,0x3daf,0x1112 ,0x2c21,0x2e5a ,
-  0x3612,0x223d ,0x3d78,0x11d3 ,0x2a65,0x2ff2 ,
-  0x3537,0x238e ,0x3d3f,0x1294 ,0x289a,0x3179 ,
-  0x3453,0x24da ,0x3d03,0x1354 ,0x26c1,0x32ef ,
-  0x3368,0x2620 ,0x3cc5,0x1413 ,0x24da,0x3453 ,
-  0x3274,0x2760 ,0x3c85,0x14d2 ,0x22e7,0x35a5 ,
-  0x3179,0x289a ,0x3c42,0x1590 ,0x20e7,0x36e5 ,
-  0x3076,0x29ce ,0x3bfd,0x164c ,0x1edc,0x3812 ,
-  0x2f6c,0x2afb ,0x3bb6,0x1709 ,0x1cc6,0x392b ,
-  0x2e5a,0x2c21 ,0x3b6d,0x17c4 ,0x1aa7,0x3a30 ,
-  0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
-  0x2c21,0x2e5a ,0x3ad3,0x1937 ,0x164c,0x3bfd ,
-  0x2afb,0x2f6c ,0x3a82,0x19ef ,0x1413,0x3cc5 ,
-  0x29ce,0x3076 ,0x3a30,0x1aa7 ,0x11d3,0x3d78 ,
-  0x289a,0x3179 ,0x39db,0x1b5d ,0x0f8d,0x3e15 ,
-  0x2760,0x3274 ,0x3984,0x1c12 ,0x0d41,0x3e9d ,
-  0x2620,0x3368 ,0x392b,0x1cc6 ,0x0af1,0x3f0f ,
-  0x24da,0x3453 ,0x38cf,0x1d79 ,0x089d,0x3f6b ,
-  0x238e,0x3537 ,0x3871,0x1e2b ,0x0646,0x3fb1 ,
-  0x223d,0x3612 ,0x3812,0x1edc ,0x03ed,0x3fe1 ,
-  0x20e7,0x36e5 ,0x37b0,0x1f8c ,0x0192,0x3ffb ,
-  0x1f8c,0x37b0 ,0x374b,0x203a ,0xff37,0x3fff ,
-  0x1e2b,0x3871 ,0x36e5,0x20e7 ,0xfcdc,0x3fec ,
-  0x1cc6,0x392b ,0x367d,0x2193 ,0xfa82,0x3fc4 ,
-  0x1b5d,0x39db ,0x3612,0x223d ,0xf82a,0x3f85 ,
-  0x19ef,0x3a82 ,0x35a5,0x22e7 ,0xf5d5,0x3f30 ,
-  0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
-  0x1709,0x3bb6 ,0x34c6,0x2435 ,0xf136,0x3e45 ,
-  0x1590,0x3c42 ,0x3453,0x24da ,0xeeee,0x3daf ,
-  0x1413,0x3cc5 ,0x33df,0x257e ,0xecac,0x3d03 ,
-  0x1294,0x3d3f ,0x3368,0x2620 ,0xea70,0x3c42 ,
-  0x1112,0x3daf ,0x32ef,0x26c1 ,0xe83c,0x3b6d ,
-  0x0f8d,0x3e15 ,0x3274,0x2760 ,0xe611,0x3a82 ,
-  0x0e06,0x3e72 ,0x31f8,0x27fe ,0xe3ee,0x3984 ,
-  0x0c7c,0x3ec5 ,0x3179,0x289a ,0xe1d5,0x3871 ,
-  0x0af1,0x3f0f ,0x30f9,0x2935 ,0xdfc6,0x374b ,
-  0x0964,0x3f4f ,0x3076,0x29ce ,0xddc3,0x3612 ,
-  0x07d6,0x3f85 ,0x2ff2,0x2a65 ,0xdbcb,0x34c6 ,
-  0x0646,0x3fb1 ,0x2f6c,0x2afb ,0xd9e0,0x3368 ,
-  0x04b5,0x3fd4 ,0x2ee4,0x2b8f ,0xd802,0x31f8 ,
-  0x0324,0x3fec ,0x2e5a,0x2c21 ,0xd632,0x3076 ,
-  0x0192,0x3ffb ,0x2dcf,0x2cb2 ,0xd471,0x2ee4 ,
-  0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
-  0xfe6e,0x3ffb ,0x2cb2,0x2dcf ,0xd11c,0x2b8f ,
-  0xfcdc,0x3fec ,0x2c21,0x2e5a ,0xcf8a,0x29ce ,
-  0xfb4b,0x3fd4 ,0x2b8f,0x2ee4 ,0xce08,0x27fe ,
-  0xf9ba,0x3fb1 ,0x2afb,0x2f6c ,0xcc98,0x2620 ,
-  0xf82a,0x3f85 ,0x2a65,0x2ff2 ,0xcb3a,0x2435 ,
-  0xf69c,0x3f4f ,0x29ce,0x3076 ,0xc9ee,0x223d ,
-  0xf50f,0x3f0f ,0x2935,0x30f9 ,0xc8b5,0x203a ,
-  0xf384,0x3ec5 ,0x289a,0x3179 ,0xc78f,0x1e2b ,
-  0xf1fa,0x3e72 ,0x27fe,0x31f8 ,0xc67c,0x1c12 ,
-  0xf073,0x3e15 ,0x2760,0x3274 ,0xc57e,0x19ef ,
-  0xeeee,0x3daf ,0x26c1,0x32ef ,0xc493,0x17c4 ,
-  0xed6c,0x3d3f ,0x2620,0x3368 ,0xc3be,0x1590 ,
-  0xebed,0x3cc5 ,0x257e,0x33df ,0xc2fd,0x1354 ,
-  0xea70,0x3c42 ,0x24da,0x3453 ,0xc251,0x1112 ,
-  0xe8f7,0x3bb6 ,0x2435,0x34c6 ,0xc1bb,0x0eca ,
-  0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
-  0xe611,0x3a82 ,0x22e7,0x35a5 ,0xc0d0,0x0a2b ,
-  0xe4a3,0x39db ,0x223d,0x3612 ,0xc07b,0x07d6 ,
-  0xe33a,0x392b ,0x2193,0x367d ,0xc03c,0x057e ,
-  0xe1d5,0x3871 ,0x20e7,0x36e5 ,0xc014,0x0324 ,
-  0xe074,0x37b0 ,0x203a,0x374b ,0xc001,0x00c9 ,
-  0xdf19,0x36e5 ,0x1f8c,0x37b0 ,0xc005,0xfe6e ,
-  0xddc3,0x3612 ,0x1edc,0x3812 ,0xc01f,0xfc13 ,
-  0xdc72,0x3537 ,0x1e2b,0x3871 ,0xc04f,0xf9ba ,
-  0xdb26,0x3453 ,0x1d79,0x38cf ,0xc095,0xf763 ,
-  0xd9e0,0x3368 ,0x1cc6,0x392b ,0xc0f1,0xf50f ,
-  0xd8a0,0x3274 ,0x1c12,0x3984 ,0xc163,0xf2bf ,
-  0xd766,0x3179 ,0x1b5d,0x39db ,0xc1eb,0xf073 ,
-  0xd632,0x3076 ,0x1aa7,0x3a30 ,0xc288,0xee2d ,
-  0xd505,0x2f6c ,0x19ef,0x3a82 ,0xc33b,0xebed ,
-  0xd3df,0x2e5a ,0x1937,0x3ad3 ,0xc403,0xe9b4 ,
-  0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
-  0xd1a6,0x2c21 ,0x17c4,0x3b6d ,0xc5d0,0xe559 ,
-  0xd094,0x2afb ,0x1709,0x3bb6 ,0xc6d5,0xe33a ,
-  0xcf8a,0x29ce ,0x164c,0x3bfd ,0xc7ee,0xe124 ,
-  0xce87,0x289a ,0x1590,0x3c42 ,0xc91b,0xdf19 ,
-  0xcd8c,0x2760 ,0x14d2,0x3c85 ,0xca5b,0xdd19 ,
-  0xcc98,0x2620 ,0x1413,0x3cc5 ,0xcbad,0xdb26 ,
-  0xcbad,0x24da ,0x1354,0x3d03 ,0xcd11,0xd93f ,
-  0xcac9,0x238e ,0x1294,0x3d3f ,0xce87,0xd766 ,
-  0xc9ee,0x223d ,0x11d3,0x3d78 ,0xd00e,0xd59b ,
-  0xc91b,0x20e7 ,0x1112,0x3daf ,0xd1a6,0xd3df ,
-  0xc850,0x1f8c ,0x1050,0x3de3 ,0xd34e,0xd231 ,
-  0xc78f,0x1e2b ,0x0f8d,0x3e15 ,0xd505,0xd094 ,
-  0xc6d5,0x1cc6 ,0x0eca,0x3e45 ,0xd6cb,0xcf07 ,
-  0xc625,0x1b5d ,0x0e06,0x3e72 ,0xd8a0,0xcd8c ,
-  0xc57e,0x19ef ,0x0d41,0x3e9d ,0xda82,0xcc21 ,
-  0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
-  0xc44a,0x1709 ,0x0bb7,0x3eeb ,0xde6d,0xc983 ,
-  0xc3be,0x1590 ,0x0af1,0x3f0f ,0xe074,0xc850 ,
-  0xc33b,0x1413 ,0x0a2b,0x3f30 ,0xe287,0xc731 ,
-  0xc2c1,0x1294 ,0x0964,0x3f4f ,0xe4a3,0xc625 ,
-  0xc251,0x1112 ,0x089d,0x3f6b ,0xe6c9,0xc52d ,
-  0xc1eb,0x0f8d ,0x07d6,0x3f85 ,0xe8f7,0xc44a ,
-  0xc18e,0x0e06 ,0x070e,0x3f9c ,0xeb2e,0xc37b ,
-  0xc13b,0x0c7c ,0x0646,0x3fb1 ,0xed6c,0xc2c1 ,
-  0xc0f1,0x0af1 ,0x057e,0x3fc4 ,0xefb0,0xc21d ,
-  0xc0b1,0x0964 ,0x04b5,0x3fd4 ,0xf1fa,0xc18e ,
-  0xc07b,0x07d6 ,0x03ed,0x3fe1 ,0xf449,0xc115 ,
-  0xc04f,0x0646 ,0x0324,0x3fec ,0xf69c,0xc0b1 ,
-  0xc02c,0x04b5 ,0x025b,0x3ff5 ,0xf8f2,0xc064 ,
-  0xc014,0x0324 ,0x0192,0x3ffb ,0xfb4b,0xc02c ,
-  0xc005,0x0192 ,0x00c9,0x3fff ,0xfda5,0xc00b ,
-  0x4000,0x0000 ,0x4000,0x0065 ,0x3fff,0x00c9 ,
-  0x3ffd,0x012e ,0x3ffb,0x0192 ,0x3ff8,0x01f7 ,
-  0x3ff5,0x025b ,0x3ff1,0x02c0 ,0x3fec,0x0324 ,
-  0x3fe7,0x0388 ,0x3fe1,0x03ed ,0x3fdb,0x0451 ,
-  0x3fd4,0x04b5 ,0x3fcc,0x051a ,0x3fc4,0x057e ,
-  0x3fbb,0x05e2 ,0x3fb1,0x0646 ,0x3fa7,0x06aa ,
-  0x3f9c,0x070e ,0x3f91,0x0772 ,0x3f85,0x07d6 ,
-  0x3f78,0x0839 ,0x3f6b,0x089d ,0x3f5d,0x0901 ,
-  0x3f4f,0x0964 ,0x3f40,0x09c7 ,0x3f30,0x0a2b ,
-  0x3f20,0x0a8e ,0x3f0f,0x0af1 ,0x3efd,0x0b54 ,
-  0x3eeb,0x0bb7 ,0x3ed8,0x0c1a ,0x3ec5,0x0c7c ,
-  0x3eb1,0x0cdf ,0x3e9d,0x0d41 ,0x3e88,0x0da4 ,
-  0x3e72,0x0e06 ,0x3e5c,0x0e68 ,0x3e45,0x0eca ,
-  0x3e2d,0x0f2b ,0x3e15,0x0f8d ,0x3dfc,0x0fee ,
-  0x3de3,0x1050 ,0x3dc9,0x10b1 ,0x3daf,0x1112 ,
-  0x3d93,0x1173 ,0x3d78,0x11d3 ,0x3d5b,0x1234 ,
-  0x3d3f,0x1294 ,0x3d21,0x12f4 ,0x3d03,0x1354 ,
-  0x3ce4,0x13b4 ,0x3cc5,0x1413 ,0x3ca5,0x1473 ,
-  0x3c85,0x14d2 ,0x3c64,0x1531 ,0x3c42,0x1590 ,
-  0x3c20,0x15ee ,0x3bfd,0x164c ,0x3bda,0x16ab ,
-  0x3bb6,0x1709 ,0x3b92,0x1766 ,0x3b6d,0x17c4 ,
-  0x3b47,0x1821 ,0x3b21,0x187e ,0x3afa,0x18db ,
-  0x3ad3,0x1937 ,0x3aab,0x1993 ,0x3a82,0x19ef ,
-  0x3a59,0x1a4b ,0x3a30,0x1aa7 ,0x3a06,0x1b02 ,
-  0x39db,0x1b5d ,0x39b0,0x1bb8 ,0x3984,0x1c12 ,
-  0x3958,0x1c6c ,0x392b,0x1cc6 ,0x38fd,0x1d20 ,
-  0x38cf,0x1d79 ,0x38a1,0x1dd3 ,0x3871,0x1e2b ,
-  0x3842,0x1e84 ,0x3812,0x1edc ,0x37e1,0x1f34 ,
-  0x37b0,0x1f8c ,0x377e,0x1fe3 ,0x374b,0x203a ,
-  0x3718,0x2091 ,0x36e5,0x20e7 ,0x36b1,0x213d ,
-  0x367d,0x2193 ,0x3648,0x21e8 ,0x3612,0x223d ,
-  0x35dc,0x2292 ,0x35a5,0x22e7 ,0x356e,0x233b ,
-  0x3537,0x238e ,0x34ff,0x23e2 ,0x34c6,0x2435 ,
-  0x348d,0x2488 ,0x3453,0x24da ,0x3419,0x252c ,
-  0x33df,0x257e ,0x33a3,0x25cf ,0x3368,0x2620 ,
-  0x332c,0x2671 ,0x32ef,0x26c1 ,0x32b2,0x2711 ,
-  0x3274,0x2760 ,0x3236,0x27af ,0x31f8,0x27fe ,
-  0x31b9,0x284c ,0x3179,0x289a ,0x3139,0x28e7 ,
-  0x30f9,0x2935 ,0x30b8,0x2981 ,0x3076,0x29ce ,
-  0x3034,0x2a1a ,0x2ff2,0x2a65 ,0x2faf,0x2ab0 ,
-  0x2f6c,0x2afb ,0x2f28,0x2b45 ,0x2ee4,0x2b8f ,
-  0x2e9f,0x2bd8 ,0x2e5a,0x2c21 ,0x2e15,0x2c6a ,
-  0x2dcf,0x2cb2 ,0x2d88,0x2cfa ,0x2d41,0x2d41 ,
-  0x2cfa,0x2d88 ,0x2cb2,0x2dcf ,0x2c6a,0x2e15 ,
-  0x2c21,0x2e5a ,0x2bd8,0x2e9f ,0x2b8f,0x2ee4 ,
-  0x2b45,0x2f28 ,0x2afb,0x2f6c ,0x2ab0,0x2faf ,
-  0x2a65,0x2ff2 ,0x2a1a,0x3034 ,0x29ce,0x3076 ,
-  0x2981,0x30b8 ,0x2935,0x30f9 ,0x28e7,0x3139 ,
-  0x289a,0x3179 ,0x284c,0x31b9 ,0x27fe,0x31f8 ,
-  0x27af,0x3236 ,0x2760,0x3274 ,0x2711,0x32b2 ,
-  0x26c1,0x32ef ,0x2671,0x332c ,0x2620,0x3368 ,
-  0x25cf,0x33a3 ,0x257e,0x33df ,0x252c,0x3419 ,
-  0x24da,0x3453 ,0x2488,0x348d ,0x2435,0x34c6 ,
-  0x23e2,0x34ff ,0x238e,0x3537 ,0x233b,0x356e ,
-  0x22e7,0x35a5 ,0x2292,0x35dc ,0x223d,0x3612 ,
-  0x21e8,0x3648 ,0x2193,0x367d ,0x213d,0x36b1 ,
-  0x20e7,0x36e5 ,0x2091,0x3718 ,0x203a,0x374b ,
-  0x1fe3,0x377e ,0x1f8c,0x37b0 ,0x1f34,0x37e1 ,
-  0x1edc,0x3812 ,0x1e84,0x3842 ,0x1e2b,0x3871 ,
-  0x1dd3,0x38a1 ,0x1d79,0x38cf ,0x1d20,0x38fd ,
-  0x1cc6,0x392b ,0x1c6c,0x3958 ,0x1c12,0x3984 ,
-  0x1bb8,0x39b0 ,0x1b5d,0x39db ,0x1b02,0x3a06 ,
-  0x1aa7,0x3a30 ,0x1a4b,0x3a59 ,0x19ef,0x3a82 ,
-  0x1993,0x3aab ,0x1937,0x3ad3 ,0x18db,0x3afa ,
-  0x187e,0x3b21 ,0x1821,0x3b47 ,0x17c4,0x3b6d ,
-  0x1766,0x3b92 ,0x1709,0x3bb6 ,0x16ab,0x3bda ,
-  0x164c,0x3bfd ,0x15ee,0x3c20 ,0x1590,0x3c42 ,
-  0x1531,0x3c64 ,0x14d2,0x3c85 ,0x1473,0x3ca5 ,
-  0x1413,0x3cc5 ,0x13b4,0x3ce4 ,0x1354,0x3d03 ,
-  0x12f4,0x3d21 ,0x1294,0x3d3f ,0x1234,0x3d5b ,
-  0x11d3,0x3d78 ,0x1173,0x3d93 ,0x1112,0x3daf ,
-  0x10b1,0x3dc9 ,0x1050,0x3de3 ,0x0fee,0x3dfc ,
-  0x0f8d,0x3e15 ,0x0f2b,0x3e2d ,0x0eca,0x3e45 ,
-  0x0e68,0x3e5c ,0x0e06,0x3e72 ,0x0da4,0x3e88 ,
-  0x0d41,0x3e9d ,0x0cdf,0x3eb1 ,0x0c7c,0x3ec5 ,
-  0x0c1a,0x3ed8 ,0x0bb7,0x3eeb ,0x0b54,0x3efd ,
-  0x0af1,0x3f0f ,0x0a8e,0x3f20 ,0x0a2b,0x3f30 ,
-  0x09c7,0x3f40 ,0x0964,0x3f4f ,0x0901,0x3f5d ,
-  0x089d,0x3f6b ,0x0839,0x3f78 ,0x07d6,0x3f85 ,
-  0x0772,0x3f91 ,0x070e,0x3f9c ,0x06aa,0x3fa7 ,
-  0x0646,0x3fb1 ,0x05e2,0x3fbb ,0x057e,0x3fc4 ,
-  0x051a,0x3fcc ,0x04b5,0x3fd4 ,0x0451,0x3fdb ,
-  0x03ed,0x3fe1 ,0x0388,0x3fe7 ,0x0324,0x3fec ,
-  0x02c0,0x3ff1 ,0x025b,0x3ff5 ,0x01f7,0x3ff8 ,
-  0x0192,0x3ffb ,0x012e,0x3ffd ,0x00c9,0x3fff ,
-  0x0065,0x4000 ,0x0000,0x4000 ,0xff9b,0x4000 ,
-  0xff37,0x3fff ,0xfed2,0x3ffd ,0xfe6e,0x3ffb ,
-  0xfe09,0x3ff8 ,0xfda5,0x3ff5 ,0xfd40,0x3ff1 ,
-  0xfcdc,0x3fec ,0xfc78,0x3fe7 ,0xfc13,0x3fe1 ,
-  0xfbaf,0x3fdb ,0xfb4b,0x3fd4 ,0xfae6,0x3fcc ,
-  0xfa82,0x3fc4 ,0xfa1e,0x3fbb ,0xf9ba,0x3fb1 ,
-  0xf956,0x3fa7 ,0xf8f2,0x3f9c ,0xf88e,0x3f91 ,
-  0xf82a,0x3f85 ,0xf7c7,0x3f78 ,0xf763,0x3f6b ,
-  0xf6ff,0x3f5d ,0xf69c,0x3f4f ,0xf639,0x3f40 ,
-  0xf5d5,0x3f30 ,0xf572,0x3f20 ,0xf50f,0x3f0f ,
-  0xf4ac,0x3efd ,0xf449,0x3eeb ,0xf3e6,0x3ed8 ,
-  0xf384,0x3ec5 ,0xf321,0x3eb1 ,0xf2bf,0x3e9d ,
-  0xf25c,0x3e88 ,0xf1fa,0x3e72 ,0xf198,0x3e5c ,
-  0xf136,0x3e45 ,0xf0d5,0x3e2d ,0xf073,0x3e15 ,
-  0xf012,0x3dfc ,0xefb0,0x3de3 ,0xef4f,0x3dc9 ,
-  0xeeee,0x3daf ,0xee8d,0x3d93 ,0xee2d,0x3d78 ,
-  0xedcc,0x3d5b ,0xed6c,0x3d3f ,0xed0c,0x3d21 ,
-  0xecac,0x3d03 ,0xec4c,0x3ce4 ,0xebed,0x3cc5 ,
-  0xeb8d,0x3ca5 ,0xeb2e,0x3c85 ,0xeacf,0x3c64 ,
-  0xea70,0x3c42 ,0xea12,0x3c20 ,0xe9b4,0x3bfd ,
-  0xe955,0x3bda ,0xe8f7,0x3bb6 ,0xe89a,0x3b92 ,
-  0xe83c,0x3b6d ,0xe7df,0x3b47 ,0xe782,0x3b21 ,
-  0xe725,0x3afa ,0xe6c9,0x3ad3 ,0xe66d,0x3aab ,
-  0xe611,0x3a82 ,0xe5b5,0x3a59 ,0xe559,0x3a30 ,
-  0xe4fe,0x3a06 ,0xe4a3,0x39db ,0xe448,0x39b0 ,
-  0xe3ee,0x3984 ,0xe394,0x3958 ,0xe33a,0x392b ,
-  0xe2e0,0x38fd ,0xe287,0x38cf ,0xe22d,0x38a1 ,
-  0xe1d5,0x3871 ,0xe17c,0x3842 ,0xe124,0x3812 ,
-  0xe0cc,0x37e1 ,0xe074,0x37b0 ,0xe01d,0x377e ,
-  0xdfc6,0x374b ,0xdf6f,0x3718 ,0xdf19,0x36e5 ,
-  0xdec3,0x36b1 ,0xde6d,0x367d ,0xde18,0x3648 ,
-  0xddc3,0x3612 ,0xdd6e,0x35dc ,0xdd19,0x35a5 ,
-  0xdcc5,0x356e ,0xdc72,0x3537 ,0xdc1e,0x34ff ,
-  0xdbcb,0x34c6 ,0xdb78,0x348d ,0xdb26,0x3453 ,
-  0xdad4,0x3419 ,0xda82,0x33df ,0xda31,0x33a3 ,
-  0xd9e0,0x3368 ,0xd98f,0x332c ,0xd93f,0x32ef ,
-  0xd8ef,0x32b2 ,0xd8a0,0x3274 ,0xd851,0x3236 ,
-  0xd802,0x31f8 ,0xd7b4,0x31b9 ,0xd766,0x3179 ,
-  0xd719,0x3139 ,0xd6cb,0x30f9 ,0xd67f,0x30b8 ,
-  0xd632,0x3076 ,0xd5e6,0x3034 ,0xd59b,0x2ff2 ,
-  0xd550,0x2faf ,0xd505,0x2f6c ,0xd4bb,0x2f28 ,
-  0xd471,0x2ee4 ,0xd428,0x2e9f ,0xd3df,0x2e5a ,
-  0xd396,0x2e15 ,0xd34e,0x2dcf ,0xd306,0x2d88 ,
-  0xd2bf,0x2d41 ,0xd278,0x2cfa ,0xd231,0x2cb2 ,
-  0xd1eb,0x2c6a ,0xd1a6,0x2c21 ,0xd161,0x2bd8 ,
-  0xd11c,0x2b8f ,0xd0d8,0x2b45 ,0xd094,0x2afb ,
-  0xd051,0x2ab0 ,0xd00e,0x2a65 ,0xcfcc,0x2a1a ,
-  0xcf8a,0x29ce ,0xcf48,0x2981 ,0xcf07,0x2935 ,
-  0xcec7,0x28e7 ,0xce87,0x289a ,0xce47,0x284c ,
-  0xce08,0x27fe ,0xcdca,0x27af ,0xcd8c,0x2760 ,
-  0xcd4e,0x2711 ,0xcd11,0x26c1 ,0xccd4,0x2671 ,
-  0xcc98,0x2620 ,0xcc5d,0x25cf ,0xcc21,0x257e ,
-  0xcbe7,0x252c ,0xcbad,0x24da ,0xcb73,0x2488 ,
-  0xcb3a,0x2435 ,0xcb01,0x23e2 ,0xcac9,0x238e ,
-  0xca92,0x233b ,0xca5b,0x22e7 ,0xca24,0x2292 ,
-  0xc9ee,0x223d ,0xc9b8,0x21e8 ,0xc983,0x2193 ,
-  0xc94f,0x213d ,0xc91b,0x20e7 ,0xc8e8,0x2091 ,
-  0xc8b5,0x203a ,0xc882,0x1fe3 ,0xc850,0x1f8c ,
-  0xc81f,0x1f34 ,0xc7ee,0x1edc ,0xc7be,0x1e84 ,
-  0xc78f,0x1e2b ,0xc75f,0x1dd3 ,0xc731,0x1d79 ,
-  0xc703,0x1d20 ,0xc6d5,0x1cc6 ,0xc6a8,0x1c6c ,
-  0xc67c,0x1c12 ,0xc650,0x1bb8 ,0xc625,0x1b5d ,
-  0xc5fa,0x1b02 ,0xc5d0,0x1aa7 ,0xc5a7,0x1a4b ,
-  0xc57e,0x19ef ,0xc555,0x1993 ,0xc52d,0x1937 ,
-  0xc506,0x18db ,0xc4df,0x187e ,0xc4b9,0x1821 ,
-  0xc493,0x17c4 ,0xc46e,0x1766 ,0xc44a,0x1709 ,
-  0xc426,0x16ab ,0xc403,0x164c ,0xc3e0,0x15ee ,
-  0xc3be,0x1590 ,0xc39c,0x1531 ,0xc37b,0x14d2 ,
-  0xc35b,0x1473 ,0xc33b,0x1413 ,0xc31c,0x13b4 ,
-  0xc2fd,0x1354 ,0xc2df,0x12f4 ,0xc2c1,0x1294 ,
-  0xc2a5,0x1234 ,0xc288,0x11d3 ,0xc26d,0x1173 ,
-  0xc251,0x1112 ,0xc237,0x10b1 ,0xc21d,0x1050 ,
-  0xc204,0x0fee ,0xc1eb,0x0f8d ,0xc1d3,0x0f2b ,
-  0xc1bb,0x0eca ,0xc1a4,0x0e68 ,0xc18e,0x0e06 ,
-  0xc178,0x0da4 ,0xc163,0x0d41 ,0xc14f,0x0cdf ,
-  0xc13b,0x0c7c ,0xc128,0x0c1a ,0xc115,0x0bb7 ,
-  0xc103,0x0b54 ,0xc0f1,0x0af1 ,0xc0e0,0x0a8e ,
-  0xc0d0,0x0a2b ,0xc0c0,0x09c7 ,0xc0b1,0x0964 ,
-  0xc0a3,0x0901 ,0xc095,0x089d ,0xc088,0x0839 ,
-  0xc07b,0x07d6 ,0xc06f,0x0772 ,0xc064,0x070e ,
-  0xc059,0x06aa ,0xc04f,0x0646 ,0xc045,0x05e2 ,
-  0xc03c,0x057e ,0xc034,0x051a ,0xc02c,0x04b5 ,
-  0xc025,0x0451 ,0xc01f,0x03ed ,0xc019,0x0388 ,
-  0xc014,0x0324 ,0xc00f,0x02c0 ,0xc00b,0x025b ,
-  0xc008,0x01f7 ,0xc005,0x0192 ,0xc003,0x012e ,
-  0xc001,0x00c9 ,0xc000,0x0065 };

src/common_audio/signal_processing_library/main/source/webrtc_fft_t_rad.c

@@ -1,27 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the Q14 radix-2 tables used in ARM9E optimization routines.
- *
- */
-
-extern const unsigned short t_Q14S_rad8[2];
-const unsigned short t_Q14S_rad8[2] = {  0x0000,0x2d41 };
-
-//extern const int t_Q30S_rad8[2];
-//const int t_Q30S_rad8[2] = {  0x00000000,0x2d413ccd };
-
-extern const unsigned short t_Q14R_rad8[2];
-const unsigned short t_Q14R_rad8[2] = {  0x2d41,0x2d41 };
-
-//extern const int t_Q30R_rad8[2];
-//const int t_Q30R_rad8[2] = {  0x2d413ccd,0x2d413ccd };

src/common_audio/signal_processing_library/main/test/unit_test/unit_test.cc

@@ -1,494 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the SPL unit_test.
- *
- */
-
-#include "unit_test.h"
-#include "signal_processing_library.h"
-
-class SplEnvironment : public ::testing::Environment {
- public:
-  virtual void SetUp() {
-  }
-  virtual void TearDown() {
-  }
-};
-
-SplTest::SplTest()
-{
-}
-
-void SplTest::SetUp() {
-}
-
-void SplTest::TearDown() {
-}
-
-TEST_F(SplTest, MacroTest) {
-    // Macros with inputs.
-    int A = 10;
-    int B = 21;
-    int a = -3;
-    int b = WEBRTC_SPL_WORD32_MAX;
-    int nr = 2;
-    int d_ptr1 = 0;
-    int d_ptr2 = 0;
-
-    EXPECT_EQ(10, WEBRTC_SPL_MIN(A, B));
-    EXPECT_EQ(21, WEBRTC_SPL_MAX(A, B));
-
-    EXPECT_EQ(3, WEBRTC_SPL_ABS_W16(a));
-    EXPECT_EQ(3, WEBRTC_SPL_ABS_W32(a));
-    EXPECT_EQ(0, WEBRTC_SPL_GET_BYTE(&B, nr));
-    WEBRTC_SPL_SET_BYTE(&d_ptr2, 1, nr);
-    EXPECT_EQ(65536, d_ptr2);
-
-    EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
-    EXPECT_EQ(-2147483645, WEBRTC_SPL_MUL(a, b));
-    EXPECT_EQ(-2147483645, WEBRTC_SPL_UMUL(a, b));
-    b = WEBRTC_SPL_WORD16_MAX >> 1;
-    EXPECT_EQ(65535, WEBRTC_SPL_UMUL_RSFT16(a, b));
-    EXPECT_EQ(1073627139, WEBRTC_SPL_UMUL_16_16(a, b));
-    EXPECT_EQ(16382, WEBRTC_SPL_UMUL_16_16_RSFT16(a, b));
-    EXPECT_EQ(-49149, WEBRTC_SPL_UMUL_32_16(a, b));
-    EXPECT_EQ(65535, WEBRTC_SPL_UMUL_32_16_RSFT16(a, b));
-    EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));
-
-    a = b;
-    b = -3;
-    EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b));
-    EXPECT_EQ(0, WEBRTC_SPL_UDIV(a, b));
-
-    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
-    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
-    EXPECT_EQ(-3, WEBRTC_SPL_MUL_16_32_RSFT14(a, b));
-    EXPECT_EQ(-24, WEBRTC_SPL_MUL_16_32_RSFT11(a, b));
-
-    int a32 = WEBRTC_SPL_WORD32_MAX;
-    int a32a = (WEBRTC_SPL_WORD32_MAX >> 16);
-    int a32b = (WEBRTC_SPL_WORD32_MAX & 0x0000ffff);
-    EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, A));
-    EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, A));
-
-    EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_16(a, b));
-    EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2));
-
-    EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2));
-    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b));
-
-    EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W32(a, b));
-    EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
-    EXPECT_EQ(21, WEBRTC_SPL_SAT(a, B, A));
-    EXPECT_EQ(-49149, WEBRTC_SPL_MUL_32_16(a, b));
-
-    EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W32(a, b));
-    EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W16(a, b));
-    EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W16(a, b));
-
-    EXPECT_TRUE(WEBRTC_SPL_IS_NEG(b));
-
-    // Shifting with negative numbers allowed
-    // Positive means left shift
-    EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W16(a, 1));
-    EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W32(a, 1));
-
-    // Shifting with negative numbers not allowed
-    // We cannot do casting here due to signed/unsigned problem
-    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W16(a, 1));
-    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W16(a, 1));
-    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W32(a, 1));
-    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
-
-    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_U16(a, 1));
-    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_U16(a, 1));
-    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_U32(a, 1));
-    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_U32(a, 1));
-
-    EXPECT_EQ(1470, WEBRTC_SPL_RAND(A));
-}
-
-TEST_F(SplTest, InlineTest) {
-
-    WebRtc_Word16 a = 121;
-    WebRtc_Word16 b = -17;
-    WebRtc_Word32 A = 111121;
-    WebRtc_Word32 B = -1711;
-    char bVersion[8];
-
-    EXPECT_EQ(104, WebRtcSpl_AddSatW16(a, b));
-    EXPECT_EQ(138, WebRtcSpl_SubSatW16(a, b));
-
-    EXPECT_EQ(109410, WebRtcSpl_AddSatW32(A, B));
-    EXPECT_EQ(112832, WebRtcSpl_SubSatW32(A, B));
-
-    EXPECT_EQ(17, WebRtcSpl_GetSizeInBits(A));
-    EXPECT_EQ(14, WebRtcSpl_NormW32(A));
-    EXPECT_EQ(4, WebRtcSpl_NormW16(B));
-    EXPECT_EQ(15, WebRtcSpl_NormU32(A));
-
-    EXPECT_EQ(0, WebRtcSpl_get_version(bVersion, 8));
-}
-
-TEST_F(SplTest, MathOperationsTest) {
-
-    int A = 117;
-    WebRtc_Word32 num = 117;
-    WebRtc_Word32 den = -5;
-    WebRtc_UWord16 denU = 5;
-    EXPECT_EQ(10, WebRtcSpl_Sqrt(A));
-    EXPECT_EQ(10, WebRtcSpl_SqrtFloor(A));
-
-
-    EXPECT_EQ(-91772805, WebRtcSpl_DivResultInQ31(den, num));
-    EXPECT_EQ(-23, WebRtcSpl_DivW32W16ResW16(num, (WebRtc_Word16)den));
-    EXPECT_EQ(-23, WebRtcSpl_DivW32W16(num, (WebRtc_Word16)den));
-    EXPECT_EQ(23, WebRtcSpl_DivU32U16(num, denU));
-    EXPECT_EQ(0, WebRtcSpl_DivW32HiLow(128, 0, 256));
-}
-
-TEST_F(SplTest, BasicArrayOperationsTest) {
-
-
-    const int kVectorSize = 4;
-    int B[] = {4, 12, 133, 1100};
-    int Bs[] = {2, 6, 66, 550};
-    WebRtc_UWord8 b8[kVectorSize];
-    WebRtc_Word16 b16[kVectorSize];
-    WebRtc_Word32 b32[kVectorSize];
-
-    WebRtc_UWord8 bTmp8[kVectorSize];
-    WebRtc_Word16 bTmp16[kVectorSize];
-    WebRtc_Word32 bTmp32[kVectorSize];
-
-    WebRtcSpl_MemSetW16(b16, 3, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(3, b16[kk]);
-    }
-    EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW16(b16, kVectorSize));
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(0, b16[kk]);
-    }
-    EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW16(b16, kVectorSize));
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(1, b16[kk]);
-    }
-    WebRtcSpl_MemSetW32(b32, 3, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(3, b32[kk]);
-    }
-    EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW32(b32, kVectorSize));
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(0, b32[kk]);
-    }
-    EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW32(b32, kVectorSize));
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(1, b32[kk]);
-    }
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        bTmp8[kk] = (WebRtc_Word8)kk;
-        bTmp16[kk] = (WebRtc_Word16)kk;
-        bTmp32[kk] = (WebRtc_Word32)kk;
-    }
-    WEBRTC_SPL_MEMCPY_W8(b8, bTmp8, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(b8[kk], bTmp8[kk]);
-    }
-    WEBRTC_SPL_MEMCPY_W16(b16, bTmp16, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(b16[kk], bTmp16[kk]);
-    }
-//    WEBRTC_SPL_MEMCPY_W32(b32, bTmp32, kVectorSize);
-//    for (int kk = 0; kk < kVectorSize; ++kk) {
-//        EXPECT_EQ(b32[kk], bTmp32[kk]);
-//    }
-    EXPECT_EQ(2, WebRtcSpl_CopyFromEndW16(b16, kVectorSize, 2, bTmp16));
-    for (int kk = 0; kk < 2; ++kk) {
-        EXPECT_EQ(kk+2, bTmp16[kk]);
-    }
-
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        b32[kk] = B[kk];
-        b16[kk] = (WebRtc_Word16)B[kk];
-    }
-    WebRtcSpl_VectorBitShiftW32ToW16(bTmp16, kVectorSize, b32, 1);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
-    }
-    WebRtcSpl_VectorBitShiftW16(bTmp16, kVectorSize, b16, 1);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
-    }
-    WebRtcSpl_VectorBitShiftW32(bTmp32, kVectorSize, b32, 1);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ((B[kk]>>1), bTmp32[kk]);
-    }
-
-    WebRtcSpl_MemCpyReversedOrder(&bTmp16[3], b16, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(b16[3-kk], bTmp16[kk]);
-    }
-
-}
-
-TEST_F(SplTest, MinMaxOperationsTest) {
-
-
-    const int kVectorSize = 4;
-    int B[] = {4, 12, 133, -1100};
-    WebRtc_Word16 b16[kVectorSize];
-    WebRtc_Word32 b32[kVectorSize];
-
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        b16[kk] = B[kk];
-        b32[kk] = B[kk];
-    }
-
-    EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW16(b16, kVectorSize));
-    EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW32(b32, kVectorSize));
-    EXPECT_EQ(133, WebRtcSpl_MaxValueW16(b16, kVectorSize));
-    EXPECT_EQ(133, WebRtcSpl_MaxValueW32(b32, kVectorSize));
-    EXPECT_EQ(3, WebRtcSpl_MaxAbsIndexW16(b16, kVectorSize));
-    EXPECT_EQ(2, WebRtcSpl_MaxIndexW16(b16, kVectorSize));
-    EXPECT_EQ(2, WebRtcSpl_MaxIndexW32(b32, kVectorSize));
-
-    EXPECT_EQ(-1100, WebRtcSpl_MinValueW16(b16, kVectorSize));
-    EXPECT_EQ(-1100, WebRtcSpl_MinValueW32(b32, kVectorSize));
-    EXPECT_EQ(3, WebRtcSpl_MinIndexW16(b16, kVectorSize));
-    EXPECT_EQ(3, WebRtcSpl_MinIndexW32(b32, kVectorSize));
-
-    EXPECT_EQ(0, WebRtcSpl_GetScalingSquare(b16, kVectorSize, 1));
-
-}
-
-TEST_F(SplTest, VectorOperationsTest) {
-
-
-    const int kVectorSize = 4;
-    int B[] = {4, 12, 133, 1100};
-    WebRtc_Word16 a16[kVectorSize];
-    WebRtc_Word16 b16[kVectorSize];
-    WebRtc_Word32 b32[kVectorSize];
-    WebRtc_Word16 bTmp16[kVectorSize];
-
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        a16[kk] = B[kk];
-        b16[kk] = B[kk];
-    }
-
-    WebRtcSpl_AffineTransformVector(bTmp16, b16, 3, 7, 2, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ((B[kk]*3+7)>>2, bTmp16[kk]);
-    }
-    WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ((B[kk]*3+B[kk]*2+2)>>2, bTmp16[kk]);
-    }
-
-    WebRtcSpl_AddAffineVectorToVector(bTmp16, b16, 3, 7, 2, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(((B[kk]*3+B[kk]*2+2)>>2)+((b16[kk]*3+7)>>2), bTmp16[kk]);
-    }
-
-    WebRtcSpl_CrossCorrelation(b32, b16, bTmp16, kVectorSize, 2, 2, 0);
-    for (int kk = 0; kk < 2; ++kk) {
-        EXPECT_EQ(614236, b32[kk]);
-    }
-//    EXPECT_EQ(, WebRtcSpl_DotProduct(b16, bTmp16, 4));
-    EXPECT_EQ(306962, WebRtcSpl_DotProductWithScale(b16, b16, kVectorSize, 2));
-
-    WebRtcSpl_ScaleVector(b16, bTmp16, 13, kVectorSize, 2);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
-    }
-    WebRtcSpl_ScaleVectorWithSat(b16, bTmp16, 13, kVectorSize, 2);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
-    }
-    WebRtcSpl_ScaleAndAddVectors(a16, 13, 2, b16, 7, 2, bTmp16, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(((a16[kk]*13)>>2)+((b16[kk]*7)>>2), bTmp16[kk]);
-    }
-
-    WebRtcSpl_AddVectorsAndShift(bTmp16, a16, b16, kVectorSize, 2);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
-    }
-    WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3], kVectorSize, 2);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ((a16[kk]*b16[3-kk])>>2, bTmp16[kk]);
-    }
-    WebRtcSpl_ElementwiseVectorMult(bTmp16, a16, b16, kVectorSize, 6);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ((a16[kk]*b16[kk])>>6, bTmp16[kk]);
-    }
-
-    WebRtcSpl_SqrtOfOneMinusXSquared(b16, kVectorSize, bTmp16);
-    for (int kk = 0; kk < kVectorSize - 1; ++kk) {
-        EXPECT_EQ(32767, bTmp16[kk]);
-    }
-    EXPECT_EQ(32749, bTmp16[kVectorSize - 1]);
-}
-
-TEST_F(SplTest, EstimatorsTest) {
-
-
-    const int kVectorSize = 4;
-    int B[] = {4, 12, 133, 1100};
-    WebRtc_Word16 b16[kVectorSize];
-    WebRtc_Word32 b32[kVectorSize];
-    WebRtc_Word16 bTmp16[kVectorSize];
-
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        b16[kk] = B[kk];
-        b32[kk] = B[kk];
-    }
-
-    EXPECT_EQ(0, WebRtcSpl_LevinsonDurbin(b32, b16, bTmp16, 2));
-
-}
-
-TEST_F(SplTest, FilterTest) {
-
-
-    const int kVectorSize = 4;
-    WebRtc_Word16 A[] = {1, 2, 33, 100};
-    WebRtc_Word16 A5[] = {1, 2, 33, 100, -5};
-    WebRtc_Word16 B[] = {4, 12, 133, 110};
-    WebRtc_Word16 b16[kVectorSize];
-    WebRtc_Word16 bTmp16[kVectorSize];
-    WebRtc_Word16 bTmp16Low[kVectorSize];
-    WebRtc_Word16 bState[kVectorSize];
-    WebRtc_Word16 bStateLow[kVectorSize];
-
-    WebRtcSpl_ZerosArrayW16(bState, kVectorSize);
-    WebRtcSpl_ZerosArrayW16(bStateLow, kVectorSize);
-
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        b16[kk] = A[kk];
-    }
-
-    // MA filters
-    WebRtcSpl_FilterMAFastQ12(b16, bTmp16, B, kVectorSize, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        //EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-    }
-    // AR filters
-    WebRtcSpl_FilterARFastQ12(b16, bTmp16, A, kVectorSize, kVectorSize);
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-//        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-    }
-    EXPECT_EQ(kVectorSize, WebRtcSpl_FilterAR(A5,
-                                              5,
-                                              b16,
-                                              kVectorSize,
-                                              bState,
-                                              kVectorSize,
-                                              bStateLow,
-                                              kVectorSize,
-                                              bTmp16,
-                                              bTmp16Low,
-                                              kVectorSize));
-
-}
-
-TEST_F(SplTest, RandTest) {
-
-
-    const int kVectorSize = 4;
-    WebRtc_Word16 BU[] = {3653, 12446, 8525, 30691};
-    WebRtc_Word16 BN[] = {3459, -11689, -258, -3738};
-    WebRtc_Word16 b16[kVectorSize];
-    WebRtc_UWord32 bSeed = 100000;
-
-    EXPECT_EQ(464449057, WebRtcSpl_IncreaseSeed(&bSeed));
-    EXPECT_EQ(31565, WebRtcSpl_RandU(&bSeed));
-    EXPECT_EQ(-9786, WebRtcSpl_RandN(&bSeed));
-    EXPECT_EQ(kVectorSize, WebRtcSpl_RandUArray(b16, kVectorSize, &bSeed));
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        EXPECT_EQ(BU[kk], b16[kk]);
-    }
-}
-
-TEST_F(SplTest, SignalProcessingTest) {
-
-
-    const int kVectorSize = 4;
-    int A[] = {1, 2, 33, 100};
-    WebRtc_Word16 b16[kVectorSize];
-    WebRtc_Word32 b32[kVectorSize];
-
-    WebRtc_Word16 bTmp16[kVectorSize];
-    WebRtc_Word32 bTmp32[kVectorSize];
-
-    int bScale = 0;
-
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        b16[kk] = A[kk];
-        b32[kk] = A[kk];
-    }
-
-    EXPECT_EQ(2, WebRtcSpl_AutoCorrelation(b16, kVectorSize, 1, bTmp32, &bScale));
-    WebRtcSpl_ReflCoefToLpc(b16, kVectorSize, bTmp16);
-//    for (int kk = 0; kk < kVectorSize; ++kk) {
-//        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-//    }
-    WebRtcSpl_LpcToReflCoef(bTmp16, kVectorSize, b16);
-//    for (int kk = 0; kk < kVectorSize; ++kk) {
-//        EXPECT_EQ(a16[kk], b16[kk]);
-//    }
-    WebRtcSpl_AutoCorrToReflCoef(b32, kVectorSize, bTmp16);
-//    for (int kk = 0; kk < kVectorSize; ++kk) {
-//        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-//    }
-    WebRtcSpl_GetHanningWindow(bTmp16, kVectorSize);
-//    for (int kk = 0; kk < kVectorSize; ++kk) {
-//        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-//    }
-
-    for (int kk = 0; kk < kVectorSize; ++kk) {
-        b16[kk] = A[kk];
-    }
-    EXPECT_EQ(11094 , WebRtcSpl_Energy(b16, kVectorSize, &bScale));
-    EXPECT_EQ(0, bScale);
-}
-
-TEST_F(SplTest, FFTTest) {
-
-
-    WebRtc_Word16 B[] = {1, 2, 33, 100,
-            2, 3, 34, 101,
-            3, 4, 35, 102,
-            4, 5, 36, 103};
-
-    EXPECT_EQ(0, WebRtcSpl_ComplexFFT(B, 3, 1));
-//    for (int kk = 0; kk < 16; ++kk) {
-//        EXPECT_EQ(A[kk], B[kk]);
-//    }
-    EXPECT_EQ(0, WebRtcSpl_ComplexIFFT(B, 3, 1));
-//    for (int kk = 0; kk < 16; ++kk) {
-//        EXPECT_EQ(A[kk], B[kk]);
-//    }
-    WebRtcSpl_ComplexBitReverse(B, 3);
-    for (int kk = 0; kk < 16; ++kk) {
-        //EXPECT_EQ(A[kk], B[kk]);
-    }
-}
-
-int main(int argc, char** argv) {
-  ::testing::InitGoogleTest(&argc, argv);
-  SplEnvironment* env = new SplEnvironment;
-  ::testing::AddGlobalTestEnvironment(env);
-
-  return RUN_ALL_TESTS();
-}

src/common_audio/signal_processing_library/main/test/unit_test/unit_test.h

@@ -1,30 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This header file contains the function WebRtcSpl_CopyFromBeginU8().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#ifndef WEBRTC_SPL_UNIT_TEST_H_
-#define WEBRTC_SPL_UNIT_TEST_H_
-
-#include <gtest/gtest.h>
-
-class SplTest: public ::testing::Test
-{
-protected:
-    SplTest();
-    virtual void SetUp();
-    virtual void TearDown();
-};
-
-#endif  // WEBRTC_SPL_UNIT_TEST_H_

src/common_audio/vad/Makefile.am

@@ -1,15 +0,0 @@
-noinst_LTLIBRARIES = libvad.la
-
-libvad_la_SOURCES = main/interface/webrtc_vad.h \
-		    main/source/webrtc_vad.c \
-		    main/source/vad_core.c \
-		    main/source/vad_core.h \
-		    main/source/vad_defines.h \
-		    main/source/vad_filterbank.c \
-		    main/source/vad_filterbank.h \
-		    main/source/vad_gmm.c \
-		    main/source/vad_gmm.h \
-		    main/source/vad_sp.c \
-		    main/source/vad_sp.h
-libvad_la_CFLAGS = $(AM_CFLAGS) $(COMMON_CFLAGS) \
-		   -I$(top_srcdir)/src/common_audio/signal_processing_library/main/interface

src/common_audio/vad/main/interface/webrtc_vad.h

@@ -1,159 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the VAD API calls. Specific function calls are given below.
- */
-
-#ifndef WEBRTC_VAD_WEBRTC_VAD_H_
-#define WEBRTC_VAD_WEBRTC_VAD_H_
-
-#include "typedefs.h"
-
-typedef struct WebRtcVadInst VadInst;
-
-#ifdef __cplusplus
-extern "C"
-{
-#endif
-
-/****************************************************************************
- * WebRtcVad_get_version(...)
- *
- * This function returns the version number of the code.
- *
- * Output:
- *      - version       : Pointer to a buffer where the version info will
- *                        be stored.
- * Input:
- *      - size_bytes    : Size of the buffer.
- *
- */
-WebRtc_Word16 WebRtcVad_get_version(char *version, size_t size_bytes);
-
-/****************************************************************************
- * WebRtcVad_AssignSize(...) 
- *
- * This functions get the size needed for storing the instance for encoder
- * and decoder, respectively
- *
- * Input/Output:
- *      - size_in_bytes : Pointer to integer where the size is returned
- *
- * Return value         : 0
- */
-WebRtc_Word16 WebRtcVad_AssignSize(int *size_in_bytes);
-
-/****************************************************************************
- * WebRtcVad_Assign(...) 
- *
- * This functions Assigns memory for the instances.
- *
- * Input:
- *        - vad_inst_addr :  Address to where to assign memory
- * Output:
- *        - vad_inst      :  Pointer to the instance that should be created
- *
- * Return value           :  0 - Ok
- *                          -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Assign(VadInst **vad_inst, void *vad_inst_addr);
-
-/****************************************************************************
- * WebRtcVad_Create(...)
- *
- * This function creates an instance to the VAD structure
- *
- * Input:
- *      - vad_inst      : Pointer to VAD instance that should be created
- *
- * Output:
- *      - vad_inst      : Pointer to created VAD instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Create(VadInst **vad_inst);
-
-/****************************************************************************
- * WebRtcVad_Free(...)
- *
- * This function frees the dynamic memory of a specified VAD instance
- *
- * Input:
- *      - vad_inst      : Pointer to VAD instance that should be freed
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Free(VadInst *vad_inst);
-
-/****************************************************************************
- * WebRtcVad_Init(...)
- *
- * This function initializes a VAD instance
- *
- * Input:
- *      - vad_inst      : Instance that should be initialized
- *
- * Output:
- *      - vad_inst      : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Init(VadInst *vad_inst);
-
-/****************************************************************************
- * WebRtcVad_set_mode(...)
- *
- * This function initializes a VAD instance
- *
- * Input:
- *      - vad_inst      : VAD instance
- *      - mode          : Aggressiveness setting (0, 1, 2, or 3) 
- *
- * Output:
- *      - vad_inst      : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-WebRtc_Word16 WebRtcVad_set_mode(VadInst *vad_inst, WebRtc_Word16 mode);
-
-/****************************************************************************
- * WebRtcVad_Process(...)
- * 
- * This functions does a VAD for the inserted speech frame
- *
- * Input
- *        - vad_inst     : VAD Instance. Needs to be initiated before call.
- *        - fs           : sampling frequency (Hz): 8000, 16000, or 32000
- *        - speech_frame : Pointer to speech frame buffer
- *        - frame_length : Length of speech frame buffer in number of samples
- *
- * Output:
- *        - vad_inst     : Updated VAD instance
- *
- * Return value          :  1 - Active Voice
- *                          0 - Non-active Voice
- *                         -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Process(VadInst *vad_inst,
-                                WebRtc_Word16 fs,
-                                WebRtc_Word16 *speech_frame,
-                                WebRtc_Word16 frame_length);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_VAD_WEBRTC_VAD_H_

src/common_audio/vad/main/source/vad.gypi

@@ -1,46 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'targets': [
-    {
-      'target_name': 'vad',
-      'type': '<(library)',
-      'dependencies': [
-        'spl',
-      ],
-      'include_dirs': [
-        '../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-        ],
-      },
-      'sources': [
-        '../interface/webrtc_vad.h',
-        'webrtc_vad.c',
-        'vad_core.c',
-        'vad_core.h',
-        'vad_defines.h',
-        'vad_filterbank.c',
-        'vad_filterbank.h',
-        'vad_gmm.c',
-        'vad_gmm.h',
-        'vad_sp.c',
-        'vad_sp.h',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:

src/common_audio/vad/main/source/vad_core.c

@@ -1,723 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the core functionality in VAD.
- * For function description, see vad_core.h.
- */
-
-#include "vad_core.h"
-
-#include "signal_processing_library.h"
-#include "typedefs.h"
-#include "vad_defines.h"
-#include "vad_filterbank.h"
-#include "vad_gmm.h"
-#include "vad_sp.h"
-
-// Spectrum Weighting
-static const WebRtc_Word16 kSpectrumWeight[6] = { 6, 8, 10, 12, 14, 16 };
-static const WebRtc_Word16 kNoiseUpdateConst = 655; // Q15
-static const WebRtc_Word16 kSpeechUpdateConst = 6554; // Q15
-static const WebRtc_Word16 kBackEta = 154; // Q8
-// Minimum difference between the two models, Q5
-static const WebRtc_Word16 kMinimumDifference[6] = {
-    544, 544, 576, 576, 576, 576 };
-// Upper limit of mean value for speech model, Q7
-static const WebRtc_Word16 kMaximumSpeech[6] = {
-    11392, 11392, 11520, 11520, 11520, 11520 };
-// Minimum value for mean value
-static const WebRtc_Word16 kMinimumMean[2] = { 640, 768 };
-// Upper limit of mean value for noise model, Q7
-static const WebRtc_Word16 kMaximumNoise[6] = {
-    9216, 9088, 8960, 8832, 8704, 8576 };
-// Start values for the Gaussian models, Q7
-// Weights for the two Gaussians for the six channels (noise)
-static const WebRtc_Word16 kNoiseDataWeights[12] = {
-    34, 62, 72, 66, 53, 25, 94, 66, 56, 62, 75, 103 };
-// Weights for the two Gaussians for the six channels (speech)
-static const WebRtc_Word16 kSpeechDataWeights[12] = {
-    48, 82, 45, 87, 50, 47, 80, 46, 83, 41, 78, 81 };
-// Means for the two Gaussians for the six channels (noise)
-static const WebRtc_Word16 kNoiseDataMeans[12] = {
-    6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863, 7820, 7266, 5020, 4362 };
-// Means for the two Gaussians for the six channels (speech)
-static const WebRtc_Word16 kSpeechDataMeans[12] = {
-    8306, 10085, 10078, 11823, 11843, 6309, 9473, 9571, 10879, 7581, 8180, 7483
-};
-// Stds for the two Gaussians for the six channels (noise)
-static const WebRtc_Word16 kNoiseDataStds[12] = {
-    378, 1064, 493, 582, 688, 593, 474, 697, 475, 688, 421, 455 };
-// Stds for the two Gaussians for the six channels (speech)
-static const WebRtc_Word16 kSpeechDataStds[12] = {
-    555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540, 1079, 850 };
-
-static const int kInitCheck = 42;
-
-// Initialize VAD
-int WebRtcVad_InitCore(VadInstT *inst, short mode)
-{
-    int i;
-
-    // Initialization of struct
-    inst->vad = 1;
-    inst->frame_counter = 0;
-    inst->over_hang = 0;
-    inst->num_of_speech = 0;
-
-    // Initialization of downsampling filter state
-    inst->downsampling_filter_states[0] = 0;
-    inst->downsampling_filter_states[1] = 0;
-    inst->downsampling_filter_states[2] = 0;
-    inst->downsampling_filter_states[3] = 0;
-
-    // Read initial PDF parameters
-    for (i = 0; i < NUM_TABLE_VALUES; i++)
-    {
-        inst->noise_means[i] = kNoiseDataMeans[i];
-        inst->speech_means[i] = kSpeechDataMeans[i];
-        inst->noise_stds[i] = kNoiseDataStds[i];
-        inst->speech_stds[i] = kSpeechDataStds[i];
-    }
-
-    // Index and Minimum value vectors are initialized
-    for (i = 0; i < 16 * NUM_CHANNELS; i++)
-    {
-        inst->low_value_vector[i] = 10000;
-        inst->index_vector[i] = 0;
-    }
-
-    for (i = 0; i < 5; i++)
-    {
-        inst->upper_state[i] = 0;
-        inst->lower_state[i] = 0;
-    }
-
-    for (i = 0; i < 4; i++)
-    {
-        inst->hp_filter_state[i] = 0;
-    }
-
-    // Init mean value memory, for FindMin function
-    inst->mean_value[0] = 1600;
-    inst->mean_value[1] = 1600;
-    inst->mean_value[2] = 1600;
-    inst->mean_value[3] = 1600;
-    inst->mean_value[4] = 1600;
-    inst->mean_value[5] = 1600;
-
-    if (mode == 0)
-    {
-        // Quality mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_Q; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_Q; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_Q; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_Q; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_Q; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_Q; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_Q;
-        inst->individual[1] = INDIVIDUAL_20MS_Q;
-        inst->individual[2] = INDIVIDUAL_30MS_Q;
-
-        inst->total[0] = TOTAL_10MS_Q;
-        inst->total[1] = TOTAL_20MS_Q;
-        inst->total[2] = TOTAL_30MS_Q;
-    } else if (mode == 1)
-    {
-        // Low bitrate mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_LBR; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_LBR; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_LBR; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_LBR;
-        inst->individual[1] = INDIVIDUAL_20MS_LBR;
-        inst->individual[2] = INDIVIDUAL_30MS_LBR;
-
-        inst->total[0] = TOTAL_10MS_LBR;
-        inst->total[1] = TOTAL_20MS_LBR;
-        inst->total[2] = TOTAL_30MS_LBR;
-    } else if (mode == 2)
-    {
-        // Aggressive mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_AGG; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_AGG; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_AGG; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_AGG;
-        inst->individual[1] = INDIVIDUAL_20MS_AGG;
-        inst->individual[2] = INDIVIDUAL_30MS_AGG;
-
-        inst->total[0] = TOTAL_10MS_AGG;
-        inst->total[1] = TOTAL_20MS_AGG;
-        inst->total[2] = TOTAL_30MS_AGG;
-    } else
-    {
-        // Very aggressive mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_VAG; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_VAG; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_VAG; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_VAG;
-        inst->individual[1] = INDIVIDUAL_20MS_VAG;
-        inst->individual[2] = INDIVIDUAL_30MS_VAG;
-
-        inst->total[0] = TOTAL_10MS_VAG;
-        inst->total[1] = TOTAL_20MS_VAG;
-        inst->total[2] = TOTAL_30MS_VAG;
-    }
-
-    inst->init_flag = kInitCheck;
-
-    return 0;
-}
-
-// Set aggressiveness mode
-int WebRtcVad_set_mode_core(VadInstT *inst, short mode)
-{
-
-    if (mode == 0)
-    {
-        // Quality mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_Q; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_Q; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_Q; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_Q; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_Q; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_Q; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_Q;
-        inst->individual[1] = INDIVIDUAL_20MS_Q;
-        inst->individual[2] = INDIVIDUAL_30MS_Q;
-
-        inst->total[0] = TOTAL_10MS_Q;
-        inst->total[1] = TOTAL_20MS_Q;
-        inst->total[2] = TOTAL_30MS_Q;
-    } else if (mode == 1)
-    {
-        // Low bitrate mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_LBR; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_LBR; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_LBR; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_LBR;
-        inst->individual[1] = INDIVIDUAL_20MS_LBR;
-        inst->individual[2] = INDIVIDUAL_30MS_LBR;
-
-        inst->total[0] = TOTAL_10MS_LBR;
-        inst->total[1] = TOTAL_20MS_LBR;
-        inst->total[2] = TOTAL_30MS_LBR;
-    } else if (mode == 2)
-    {
-        // Aggressive mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_AGG; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_AGG; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_AGG; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_AGG;
-        inst->individual[1] = INDIVIDUAL_20MS_AGG;
-        inst->individual[2] = INDIVIDUAL_30MS_AGG;
-
-        inst->total[0] = TOTAL_10MS_AGG;
-        inst->total[1] = TOTAL_20MS_AGG;
-        inst->total[2] = TOTAL_30MS_AGG;
-    } else if (mode == 3)
-    {
-        // Very aggressive mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_VAG; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_VAG; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_VAG; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_VAG;
-        inst->individual[1] = INDIVIDUAL_20MS_VAG;
-        inst->individual[2] = INDIVIDUAL_30MS_VAG;
-
-        inst->total[0] = TOTAL_10MS_VAG;
-        inst->total[1] = TOTAL_20MS_VAG;
-        inst->total[2] = TOTAL_30MS_VAG;
-    } else
-    {
-        return -1;
-    }
-
-    return 0;
-}
-
-// Calculate VAD decision by first extracting feature values and then calculate
-// probability for both speech and background noise.
-
-WebRtc_Word16 WebRtcVad_CalcVad32khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
-                                     int frame_length)
-{
-    WebRtc_Word16 len, vad;
-    WebRtc_Word16 speechWB[480]; // Downsampled speech frame: 960 samples (30ms in SWB)
-    WebRtc_Word16 speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
-
-
-    // Downsample signal 32->16->8 before doing VAD
-    WebRtcVad_Downsampling(speech_frame, speechWB, &(inst->downsampling_filter_states[2]),
-                           frame_length);
-    len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
-
-    WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states, len);
-    len = WEBRTC_SPL_RSHIFT_W16(len, 1);
-
-    // Do VAD on an 8 kHz signal
-    vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
-
-    return vad;
-}
-
-WebRtc_Word16 WebRtcVad_CalcVad16khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
-                                     int frame_length)
-{
-    WebRtc_Word16 len, vad;
-    WebRtc_Word16 speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
-
-    // Wideband: Downsample signal before doing VAD
-    WebRtcVad_Downsampling(speech_frame, speechNB, inst->downsampling_filter_states,
-                           frame_length);
-
-    len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
-    vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
-
-    return vad;
-}
-
-WebRtc_Word16 WebRtcVad_CalcVad8khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
-                                    int frame_length)
-{
-    WebRtc_Word16 feature_vector[NUM_CHANNELS], total_power;
-
-    // Get power in the bands
-    total_power = WebRtcVad_get_features(inst, speech_frame, frame_length, feature_vector);
-
-    // Make a VAD
-    inst->vad = WebRtcVad_GmmProbability(inst, feature_vector, total_power, frame_length);
-
-    return inst->vad;
-}
-
-// Calculate probability for both speech and background noise, and perform a
-// hypothesis-test.
-WebRtc_Word16 WebRtcVad_GmmProbability(VadInstT *inst, WebRtc_Word16 *feature_vector,
-                                       WebRtc_Word16 total_power, int frame_length)
-{
-    int n, k;
-    WebRtc_Word16 backval;
-    WebRtc_Word16 h0, h1;
-    WebRtc_Word16 ratvec, xval;
-    WebRtc_Word16 vadflag;
-    WebRtc_Word16 shifts0, shifts1;
-    WebRtc_Word16 tmp16, tmp16_1, tmp16_2;
-    WebRtc_Word16 diff, nr, pos;
-    WebRtc_Word16 nmk, nmk2, nmk3, smk, smk2, nsk, ssk;
-    WebRtc_Word16 delt, ndelt;
-    WebRtc_Word16 maxspe, maxmu;
-    WebRtc_Word16 deltaN[NUM_TABLE_VALUES], deltaS[NUM_TABLE_VALUES];
-    WebRtc_Word16 ngprvec[NUM_TABLE_VALUES], sgprvec[NUM_TABLE_VALUES];
-    WebRtc_Word32 h0test, h1test;
-    WebRtc_Word32 tmp32_1, tmp32_2;
-    WebRtc_Word32 dotVal;
-    WebRtc_Word32 nmid, smid;
-    WebRtc_Word32 probn[NUM_MODELS], probs[NUM_MODELS];
-    WebRtc_Word16 *nmean1ptr, *nmean2ptr, *smean1ptr, *smean2ptr, *nstd1ptr, *nstd2ptr,
-            *sstd1ptr, *sstd2ptr;
-    WebRtc_Word16 overhead1, overhead2, individualTest, totalTest;
-
-    // Set the thresholds to different values based on frame length
-    if (frame_length == 80)
-    {
-        // 80 input samples
-        overhead1 = inst->over_hang_max_1[0];
-        overhead2 = inst->over_hang_max_2[0];
-        individualTest = inst->individual[0];
-        totalTest = inst->total[0];
-    } else if (frame_length == 160)
-    {
-        // 160 input samples
-        overhead1 = inst->over_hang_max_1[1];
-        overhead2 = inst->over_hang_max_2[1];
-        individualTest = inst->individual[1];
-        totalTest = inst->total[1];
-    } else
-    {
-        // 240 input samples
-        overhead1 = inst->over_hang_max_1[2];
-        overhead2 = inst->over_hang_max_2[2];
-        individualTest = inst->individual[2];
-        totalTest = inst->total[2];
-    }
-
-    if (total_power > MIN_ENERGY)
-    { // If signal present at all
-
-        // Set pointers to the gaussian parameters
-        nmean1ptr = &inst->noise_means[0];
-        nmean2ptr = &inst->noise_means[NUM_CHANNELS];
-        smean1ptr = &inst->speech_means[0];
-        smean2ptr = &inst->speech_means[NUM_CHANNELS];
-        nstd1ptr = &inst->noise_stds[0];
-        nstd2ptr = &inst->noise_stds[NUM_CHANNELS];
-        sstd1ptr = &inst->speech_stds[0];
-        sstd2ptr = &inst->speech_stds[NUM_CHANNELS];
-
-        vadflag = 0;
-        dotVal = 0;
-        for (n = 0; n < NUM_CHANNELS; n++)
-        { // For all channels
-
-            pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
-            xval = feature_vector[n];
-
-            // Probability for Noise, Q7 * Q20 = Q27
-            tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean1ptr++, *nstd1ptr++,
-                                                    &deltaN[pos]);
-            probn[0] = (WebRtc_Word32)(kNoiseDataWeights[n] * tmp32_1);
-            tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean2ptr++, *nstd2ptr++,
-                                                    &deltaN[pos + 1]);
-            probn[1] = (WebRtc_Word32)(kNoiseDataWeights[n + NUM_CHANNELS] * tmp32_1);
-            h0test = probn[0] + probn[1]; // Q27
-            h0 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(h0test, 12); // Q15
-
-            // Probability for Speech
-            tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean1ptr++, *sstd1ptr++,
-                                                    &deltaS[pos]);
-            probs[0] = (WebRtc_Word32)(kSpeechDataWeights[n] * tmp32_1);
-            tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean2ptr++, *sstd2ptr++,
-                                                    &deltaS[pos + 1]);
-            probs[1] = (WebRtc_Word32)(kSpeechDataWeights[n + NUM_CHANNELS] * tmp32_1);
-            h1test = probs[0] + probs[1]; // Q27
-            h1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(h1test, 12); // Q15
-
-            // Get likelihood ratio. Approximate log2(H1/H0) with shifts0 - shifts1
-            shifts0 = WebRtcSpl_NormW32(h0test);
-            shifts1 = WebRtcSpl_NormW32(h1test);
-
-            if ((h0test > 0) && (h1test > 0))
-            {
-                ratvec = shifts0 - shifts1;
-            } else if (h1test > 0)
-            {
-                ratvec = 31 - shifts1;
-            } else if (h0test > 0)
-            {
-                ratvec = shifts0 - 31;
-            } else
-            {
-                ratvec = 0;
-            }
-
-            // VAD decision with spectrum weighting
-            dotVal += WEBRTC_SPL_MUL_16_16(ratvec, kSpectrumWeight[n]);
-
-            // Individual channel test
-            if ((ratvec << 2) > individualTest)
-            {
-                vadflag = 1;
-            }
-
-            // Probabilities used when updating model
-            if (h0 > 0)
-            {
-                tmp32_1 = probn[0] & 0xFFFFF000; // Q27
-                tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2); // Q29
-                ngprvec[pos] = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, h0);
-                ngprvec[pos + 1] = 16384 - ngprvec[pos];
-            } else
-            {
-                ngprvec[pos] = 16384;
-                ngprvec[pos + 1] = 0;
-            }
-
-            // Probabilities used when updating model
-            if (h1 > 0)
-            {
-                tmp32_1 = probs[0] & 0xFFFFF000;
-                tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2);
-                sgprvec[pos] = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, h1);
-                sgprvec[pos + 1] = 16384 - sgprvec[pos];
-            } else
-            {
-                sgprvec[pos] = 0;
-                sgprvec[pos + 1] = 0;
-            }
-        }
-
-        // Overall test
-        if (dotVal >= totalTest)
-        {
-            vadflag |= 1;
-        }
-
-        // Set pointers to the means and standard deviations.
-        nmean1ptr = &inst->noise_means[0];
-        smean1ptr = &inst->speech_means[0];
-        nstd1ptr = &inst->noise_stds[0];
-        sstd1ptr = &inst->speech_stds[0];
-
-        maxspe = 12800;
-
-        // Update the model's parameters
-        for (n = 0; n < NUM_CHANNELS; n++)
-        {
-
-            pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
-
-            // Get min value in past which is used for long term correction
-            backval = WebRtcVad_FindMinimum(inst, feature_vector[n], n); // Q4
-
-            // Compute the "global" mean, that is the sum of the two means weighted
-            nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr); // Q7 * Q7
-            nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+NUM_CHANNELS],
-                    *(nmean1ptr+NUM_CHANNELS));
-            tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 6); // Q8
-
-            for (k = 0; k < NUM_MODELS; k++)
-            {
-
-                nr = pos + k;
-
-                nmean2ptr = nmean1ptr + k * NUM_CHANNELS;
-                smean2ptr = smean1ptr + k * NUM_CHANNELS;
-                nstd2ptr = nstd1ptr + k * NUM_CHANNELS;
-                sstd2ptr = sstd1ptr + k * NUM_CHANNELS;
-                nmk = *nmean2ptr;
-                smk = *smean2ptr;
-                nsk = *nstd2ptr;
-                ssk = *sstd2ptr;
-
-                // Update noise mean vector if the frame consists of noise only
-                nmk2 = nmk;
-                if (!vadflag)
-                {
-                    // deltaN = (x-mu)/sigma^2
-                    // ngprvec[k] = probn[k]/(probn[0] + probn[1])
-
-                    delt = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ngprvec[nr],
-                            deltaN[nr], 11); // Q14*Q11
-                    nmk2 = nmk + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delt,
-                            kNoiseUpdateConst,
-                            22); // Q7+(Q14*Q15>>22)
-                }
-
-                // Long term correction of the noise mean
-                ndelt = WEBRTC_SPL_LSHIFT_W16(backval, 4);
-                ndelt -= tmp16_1; // Q8 - Q8
-                nmk3 = nmk2 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ndelt,
-                        kBackEta,
-                        9); // Q7+(Q8*Q8)>>9
-
-                // Control that the noise mean does not drift to much
-                tmp16 = WEBRTC_SPL_LSHIFT_W16(k+5, 7);
-                if (nmk3 < tmp16)
-                    nmk3 = tmp16;
-                tmp16 = WEBRTC_SPL_LSHIFT_W16(72+k-n, 7);
-                if (nmk3 > tmp16)
-                    nmk3 = tmp16;
-                *nmean2ptr = nmk3;
-
-                if (vadflag)
-                {
-                    // Update speech mean vector:
-                    // deltaS = (x-mu)/sigma^2
-                    // sgprvec[k] = probn[k]/(probn[0] + probn[1])
-
-                    delt = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sgprvec[nr],
-                            deltaS[nr],
-                            11); // (Q14*Q11)>>11=Q14
-                    tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delt,
-                            kSpeechUpdateConst,
-                            21) + 1;
-                    smk2 = smk + (tmp16 >> 1); // Q7 + (Q14 * Q15 >> 22)
-
-                    // Control that the speech mean does not drift to much
-                    maxmu = maxspe + 640;
-                    if (smk2 < kMinimumMean[k])
-                        smk2 = kMinimumMean[k];
-                    if (smk2 > maxmu)
-                        smk2 = maxmu;
-
-                    *smean2ptr = smk2;
-
-                    // (Q7>>3) = Q4
-                    tmp16 = WEBRTC_SPL_RSHIFT_W16((smk + 4), 3);
-
-                    tmp16 = feature_vector[n] - tmp16; // Q4
-                    tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaS[nr], tmp16, 3);
-                    tmp32_2 = tmp32_1 - (WebRtc_Word32)4096; // Q12
-                    tmp16 = WEBRTC_SPL_RSHIFT_W16((sgprvec[nr]), 2);
-                    tmp32_1 = (WebRtc_Word32)(tmp16 * tmp32_2);// (Q15>>3)*(Q14>>2)=Q12*Q12=Q24
-
-                    tmp32_2 = WEBRTC_SPL_RSHIFT_W32(tmp32_1, 4); // Q20
-
-                    // 0.1 * Q20 / Q7 = Q13
-                    if (tmp32_2 > 0)
-                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, ssk * 10);
-                    else
-                    {
-                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(-tmp32_2, ssk * 10);
-                        tmp16 = -tmp16;
-                    }
-                    // divide by 4 giving an update factor of 0.025
-                    tmp16 += 128; // Rounding
-                    ssk += WEBRTC_SPL_RSHIFT_W16(tmp16, 8);
-                    // Division with 8 plus Q7
-                    if (ssk < MIN_STD)
-                        ssk = MIN_STD;
-                    *sstd2ptr = ssk;
-                } else
-                {
-                    // Update GMM variance vectors
-                    // deltaN * (feature_vector[n] - nmk) - 1, Q11 * Q4
-                    tmp16 = feature_vector[n] - WEBRTC_SPL_RSHIFT_W16(nmk, 3);
-
-                    // (Q15>>3) * (Q14>>2) = Q12 * Q12 = Q24
-                    tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaN[nr], tmp16, 3) - 4096;
-                    tmp16 = WEBRTC_SPL_RSHIFT_W16((ngprvec[nr]+2), 2);
-                    tmp32_2 = (WebRtc_Word32)(tmp16 * tmp32_1);
-                    tmp32_1 = WEBRTC_SPL_RSHIFT_W32(tmp32_2, 14);
-                    // Q20  * approx 0.001 (2^-10=0.0009766)
-
-                    // Q20 / Q7 = Q13
-                    tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_1, nsk);
-                    if (tmp32_1 > 0)
-                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_1, nsk);
-                    else
-                    {
-                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(-tmp32_1, nsk);
-                        tmp16 = -tmp16;
-                    }
-                    tmp16 += 32; // Rounding
-                    nsk += WEBRTC_SPL_RSHIFT_W16(tmp16, 6);
-
-                    if (nsk < MIN_STD)
-                        nsk = MIN_STD;
-
-                    *nstd2ptr = nsk;
-                }
-            }
-
-            // Separate models if they are too close - nmid in Q14
-            nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr);
-            nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+NUM_CHANNELS], *nmean2ptr);
-
-            // smid in Q14
-            smid = WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n], *smean1ptr);
-            smid += WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n+NUM_CHANNELS], *smean2ptr);
-
-            // diff = "global" speech mean - "global" noise mean
-            diff = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(smid, 9);
-            tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 9);
-            diff -= tmp16;
-
-            if (diff < kMinimumDifference[n])
-            {
-
-                tmp16 = kMinimumDifference[n] - diff; // Q5
-
-                // tmp16_1 = ~0.8 * (kMinimumDifference - diff) in Q7
-                // tmp16_2 = ~0.2 * (kMinimumDifference - diff) in Q7
-                tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(13, tmp16, 2);
-                tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(3, tmp16, 2);
-
-                // First Gauss, speech model
-                tmp16 = tmp16_1 + *smean1ptr;
-                *smean1ptr = tmp16;
-                smid = WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n]);
-
-                // Second Gauss, speech model
-                tmp16 = tmp16_1 + *smean2ptr;
-                *smean2ptr = tmp16;
-                smid += WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n+NUM_CHANNELS]);
-
-                // First Gauss, noise model
-                tmp16 = *nmean1ptr - tmp16_2;
-                *nmean1ptr = tmp16;
-
-                nmid = WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n]);
-
-                // Second Gauss, noise model
-                tmp16 = *nmean2ptr - tmp16_2;
-                *nmean2ptr = tmp16;
-                nmid += WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n+NUM_CHANNELS]);
-            }
-
-            // Control that the speech & noise means do not drift to much
-            maxspe = kMaximumSpeech[n];
-            tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(smid, 7);
-            if (tmp16_2 > maxspe)
-            { // Upper limit of speech model
-                tmp16_2 -= maxspe;
-
-                *smean1ptr -= tmp16_2;
-                *smean2ptr -= tmp16_2;
-            }
-
-            tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 7);
-            if (tmp16_2 > kMaximumNoise[n])
-            {
-                tmp16_2 -= kMaximumNoise[n];
-
-                *nmean1ptr -= tmp16_2;
-                *nmean2ptr -= tmp16_2;
-            }
-
-            nmean1ptr++;
-            smean1ptr++;
-            nstd1ptr++;
-            sstd1ptr++;
-        }
-        inst->frame_counter++;
-    } else
-    {
-        vadflag = 0;
-    }
-
-    // Hangover smoothing
-    if (!vadflag)
-    {
-        if (inst->over_hang > 0)
-        {
-            vadflag = 2 + inst->over_hang;
-            inst->over_hang = inst->over_hang - 1;
-        }
-        inst->num_of_speech = 0;
-    } else
-    {
-        inst->num_of_speech = inst->num_of_speech + 1;
-        if (inst->num_of_speech > NSP_MAX)
-        {
-            inst->num_of_speech = NSP_MAX;
-            inst->over_hang = overhead2;
-        } else
-            inst->over_hang = overhead1;
-    }
-    return vadflag;
-}

src/common_audio/vad/main/source/vad_core.h

@@ -1,132 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the descriptions of the core VAD calls.
- */
-
-#ifndef WEBRTC_VAD_CORE_H_
-#define WEBRTC_VAD_CORE_H_
-
-#include "typedefs.h"
-#include "vad_defines.h"
-
-typedef struct VadInstT_
-{
-
-    WebRtc_Word16 vad;
-    WebRtc_Word32 downsampling_filter_states[4];
-    WebRtc_Word16 noise_means[NUM_TABLE_VALUES];
-    WebRtc_Word16 speech_means[NUM_TABLE_VALUES];
-    WebRtc_Word16 noise_stds[NUM_TABLE_VALUES];
-    WebRtc_Word16 speech_stds[NUM_TABLE_VALUES];
-    WebRtc_Word32 frame_counter;
-    WebRtc_Word16 over_hang; // Over Hang
-    WebRtc_Word16 num_of_speech;
-    WebRtc_Word16 index_vector[16 * NUM_CHANNELS];
-    WebRtc_Word16 low_value_vector[16 * NUM_CHANNELS];
-    WebRtc_Word16 mean_value[NUM_CHANNELS];
-    WebRtc_Word16 upper_state[5];
-    WebRtc_Word16 lower_state[5];
-    WebRtc_Word16 hp_filter_state[4];
-    WebRtc_Word16 over_hang_max_1[3];
-    WebRtc_Word16 over_hang_max_2[3];
-    WebRtc_Word16 individual[3];
-    WebRtc_Word16 total[3];
-
-    short init_flag;
-
-} VadInstT;
-
-/****************************************************************************
- * WebRtcVad_InitCore(...)
- *
- * This function initializes a VAD instance
- *
- * Input:
- *      - inst      : Instance that should be initialized
- *      - mode      : Aggressiveness degree
- *                    0 (High quality) - 3 (Highly aggressive)
- *
- * Output:
- *      - inst      : Initialized instance
- *
- * Return value     :  0 - Ok
- *                    -1 - Error
- */
-int WebRtcVad_InitCore(VadInstT* inst, short mode);
-
-/****************************************************************************
- * WebRtcVad_set_mode_core(...)
- *
- * This function changes the VAD settings
- *
- * Input:
- *      - inst      : VAD instance
- *      - mode      : Aggressiveness degree
- *                    0 (High quality) - 3 (Highly aggressive)
- *
- * Output:
- *      - inst      : Changed  instance
- *
- * Return value     :  0 - Ok
- *                    -1 - Error
- */
-
-int WebRtcVad_set_mode_core(VadInstT* inst, short mode);
-
-/****************************************************************************
- * WebRtcVad_CalcVad32khz(...) 
- * WebRtcVad_CalcVad16khz(...) 
- * WebRtcVad_CalcVad8khz(...) 
- *
- * Calculate probability for active speech and make VAD decision.
- *
- * Input:
- *      - inst          : Instance that should be initialized
- *      - speech_frame  : Input speech frame
- *      - frame_length  : Number of input samples
- *
- * Output:
- *      - inst          : Updated filter states etc.
- *
- * Return value         : VAD decision
- *                        0 - No active speech
- *                        1-6 - Active speech
- */
-WebRtc_Word16 WebRtcVad_CalcVad32khz(VadInstT* inst, WebRtc_Word16* speech_frame,
-                                     int frame_length);
-WebRtc_Word16 WebRtcVad_CalcVad16khz(VadInstT* inst, WebRtc_Word16* speech_frame,
-                                     int frame_length);
-WebRtc_Word16 WebRtcVad_CalcVad8khz(VadInstT* inst, WebRtc_Word16* speech_frame,
-                                    int frame_length);
-
-/****************************************************************************
- * WebRtcVad_GmmProbability(...)
- *
- * This function calculates the probabilities for background noise and
- * speech using Gaussian Mixture Models. A hypothesis-test is performed to decide
- * which type of signal is most probable.
- *
- * Input:
- *      - inst              : Pointer to VAD instance
- *      - feature_vector    : Feature vector = log10(energy in frequency band)
- *      - total_power       : Total power in frame.
- *      - frame_length      : Number of input samples
- *
- * Output:
- *      VAD decision        : 0 - noise, 1 - speech
- *    
- */
-WebRtc_Word16 WebRtcVad_GmmProbability(VadInstT* inst, WebRtc_Word16* feature_vector,
-                                       WebRtc_Word16 total_power, int frame_length);
-
-#endif // WEBRTC_VAD_CORE_H_

src/common_audio/vad/main/source/vad_defines.h

@@ -1,95 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the macros used in VAD.
- */
-
-#ifndef WEBRTC_VAD_DEFINES_H_
-#define WEBRTC_VAD_DEFINES_H_
-
-#define NUM_CHANNELS        6   // Eight frequency bands
-#define NUM_MODELS          2   // Number of Gaussian models
-#define NUM_TABLE_VALUES    NUM_CHANNELS * NUM_MODELS
-
-#define MIN_ENERGY          10
-#define ALPHA1              6553    // 0.2 in Q15
-#define ALPHA2              32439   // 0.99 in Q15
-#define NSP_MAX             6       // Maximum number of VAD=1 frames in a row counted
-#define MIN_STD             384     // Minimum standard deviation
-// Mode 0, Quality thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_Q   24
-#define INDIVIDUAL_20MS_Q   21      // (log10(2)*66)<<2 ~=16
-#define INDIVIDUAL_30MS_Q   24
-
-#define TOTAL_10MS_Q        57
-#define TOTAL_20MS_Q        48
-#define TOTAL_30MS_Q        57
-
-#define OHMAX1_10MS_Q       8  // Max Overhang 1
-#define OHMAX2_10MS_Q       14 // Max Overhang 2
-#define OHMAX1_20MS_Q       4  // Max Overhang 1
-#define OHMAX2_20MS_Q       7  // Max Overhang 2
-#define OHMAX1_30MS_Q       3
-#define OHMAX2_30MS_Q       5
-
-// Mode 1, Low bitrate thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_LBR 37
-#define INDIVIDUAL_20MS_LBR 32
-#define INDIVIDUAL_30MS_LBR 37
-
-#define TOTAL_10MS_LBR      100
-#define TOTAL_20MS_LBR      80
-#define TOTAL_30MS_LBR      100
-
-#define OHMAX1_10MS_LBR     8  // Max Overhang 1
-#define OHMAX2_10MS_LBR     14 // Max Overhang 2
-#define OHMAX1_20MS_LBR     4
-#define OHMAX2_20MS_LBR     7
-
-#define OHMAX1_30MS_LBR     3
-#define OHMAX2_30MS_LBR     5
-
-// Mode 2, Very aggressive thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_AGG 82
-#define INDIVIDUAL_20MS_AGG 78
-#define INDIVIDUAL_30MS_AGG 82
-
-#define TOTAL_10MS_AGG      285 //580
-#define TOTAL_20MS_AGG      260
-#define TOTAL_30MS_AGG      285
-
-#define OHMAX1_10MS_AGG     6  // Max Overhang 1
-#define OHMAX2_10MS_AGG     9  // Max Overhang 2
-#define OHMAX1_20MS_AGG     3
-#define OHMAX2_20MS_AGG     5
-
-#define OHMAX1_30MS_AGG     2
-#define OHMAX2_30MS_AGG     3
-
-// Mode 3, Super aggressive thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_VAG 94
-#define INDIVIDUAL_20MS_VAG 94
-#define INDIVIDUAL_30MS_VAG 94
-
-#define TOTAL_10MS_VAG      1100 //1700
-#define TOTAL_20MS_VAG      1050
-#define TOTAL_30MS_VAG      1100
-
-#define OHMAX1_10MS_VAG     6  // Max Overhang 1
-#define OHMAX2_10MS_VAG     9  // Max Overhang 2
-#define OHMAX1_20MS_VAG     3
-#define OHMAX2_20MS_VAG     5
-
-#define OHMAX1_30MS_VAG     2
-#define OHMAX2_30MS_VAG     3
-
-#endif // WEBRTC_VAD_DEFINES_H_

src/common_audio/vad/main/source/vad_filterbank.c

@@ -1,279 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the internal filterbank associated functions.
- * For function description, see vad_filterbank.h.
- */
-
-#include "vad_filterbank.h"
-
-#include "signal_processing_library.h"
-#include "typedefs.h"
-#include "vad_defines.h"
-
-// Constant 160*log10(2) in Q9
-static const WebRtc_Word16 kLogConst = 24660;
-// Coefficients used by WebRtcVad_HpOutput, Q14
-static const WebRtc_Word16 kHpZeroCoefs[3] = {6631, -13262, 6631};
-static const WebRtc_Word16 kHpPoleCoefs[3] = {16384, -7756, 5620};
-// Allpass filter coefficients, upper and lower, in Q15
-// Upper: 0.64, Lower: 0.17
-static const WebRtc_Word16 kAllPassCoefsQ15[2] = {20972, 5571};
-// Adjustment for division with two in WebRtcVad_SplitFilter
-static const WebRtc_Word16 kOffsetVector[6] = {368, 368, 272, 176, 176, 176};
-
-void WebRtcVad_HpOutput(WebRtc_Word16 *in_vector,
-                        WebRtc_Word16 in_vector_length,
-                        WebRtc_Word16 *out_vector,
-                        WebRtc_Word16 *filter_state)
-{
-    WebRtc_Word16 i, *pi, *outPtr;
-    WebRtc_Word32 tmpW32;
-
-    pi = &in_vector[0];
-    outPtr = &out_vector[0];
-
-    // The sum of the absolute values of the impulse response:
-    // The zero/pole-filter has a max amplification of a single sample of: 1.4546
-    // Impulse response: 0.4047 -0.6179 -0.0266  0.1993  0.1035  -0.0194
-    // The all-zero section has a max amplification of a single sample of: 1.6189
-    // Impulse response: 0.4047 -0.8094  0.4047  0       0        0
-    // The all-pole section has a max amplification of a single sample of: 1.9931
-    // Impulse response: 1.0000  0.4734 -0.1189 -0.2187 -0.0627   0.04532
-
-    for (i = 0; i < in_vector_length; i++)
-    {
-        // all-zero section (filter coefficients in Q14)
-        tmpW32 = (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[0], (*pi));
-        tmpW32 += (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[1], filter_state[0]);
-        tmpW32 += (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[2], filter_state[1]); // Q14
-        filter_state[1] = filter_state[0];
-        filter_state[0] = *pi++;
-
-        // all-pole section
-        tmpW32 -= (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[1], filter_state[2]); // Q14
-        tmpW32 -= (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[2], filter_state[3]);
-        filter_state[3] = filter_state[2];
-        filter_state[2] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32 (tmpW32, 14);
-        *outPtr++ = filter_state[2];
-    }
-}
-
-void WebRtcVad_Allpass(WebRtc_Word16 *in_vector,
-                       WebRtc_Word16 *out_vector,
-                       WebRtc_Word16 filter_coefficients,
-                       int vector_length,
-                       WebRtc_Word16 *filter_state)
-{
-    // The filter can only cause overflow (in the w16 output variable)
-    // if more than 4 consecutive input numbers are of maximum value and
-    // has the the same sign as the impulse responses first taps.
-    // First 6 taps of the impulse response: 0.6399 0.5905 -0.3779
-    // 0.2418 -0.1547 0.0990
-
-    int n;
-    WebRtc_Word16 tmp16;
-    WebRtc_Word32 tmp32, in32, state32;
-
-    state32 = WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)(*filter_state)), 16); // Q31
-
-    for (n = 0; n < vector_length; n++)
-    {
-
-        tmp32 = state32 + WEBRTC_SPL_MUL_16_16(filter_coefficients, (*in_vector));
-        tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
-        *out_vector++ = tmp16;
-        in32 = WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)(*in_vector)), 14);
-        state32 = in32 - WEBRTC_SPL_MUL_16_16(filter_coefficients, tmp16);
-        state32 = WEBRTC_SPL_LSHIFT_W32(state32, 1);
-        in_vector += 2;
-    }
-
-    *filter_state = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(state32, 16);
-}
-
-void WebRtcVad_SplitFilter(WebRtc_Word16 *in_vector,
-                           WebRtc_Word16 *out_vector_hp,
-                           WebRtc_Word16 *out_vector_lp,
-                           WebRtc_Word16 *upper_state,
-                           WebRtc_Word16 *lower_state,
-                           int in_vector_length)
-{
-    WebRtc_Word16 tmpOut;
-    int k, halflen;
-
-    // Downsampling by 2 and get two branches
-    halflen = WEBRTC_SPL_RSHIFT_W16(in_vector_length, 1);
-
-    // All-pass filtering upper branch
-    WebRtcVad_Allpass(&in_vector[0], out_vector_hp, kAllPassCoefsQ15[0], halflen, upper_state);
-
-    // All-pass filtering lower branch
-    WebRtcVad_Allpass(&in_vector[1], out_vector_lp, kAllPassCoefsQ15[1], halflen, lower_state);
-
-    // Make LP and HP signals
-    for (k = 0; k < halflen; k++)
-    {
-        tmpOut = *out_vector_hp;
-        *out_vector_hp++ -= *out_vector_lp;
-        *out_vector_lp++ += tmpOut;
-    }
-}
-
-WebRtc_Word16 WebRtcVad_get_features(VadInstT *inst,
-                                     WebRtc_Word16 *in_vector,
-                                     int frame_size,
-                                     WebRtc_Word16 *out_vector)
-{
-    int curlen, filtno;
-    WebRtc_Word16 vecHP1[120], vecLP1[120];
-    WebRtc_Word16 vecHP2[60], vecLP2[60];
-    WebRtc_Word16 *ptin;
-    WebRtc_Word16 *hptout, *lptout;
-    WebRtc_Word16 power = 0;
-
-    // Split at 2000 Hz and downsample
-    filtno = 0;
-    ptin = in_vector;
-    hptout = vecHP1;
-    lptout = vecLP1;
-    curlen = frame_size;
-    WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
-                  &inst->lower_state[filtno], curlen);
-
-    // Split at 3000 Hz and downsample
-    filtno = 1;
-    ptin = vecHP1;
-    hptout = vecHP2;
-    lptout = vecLP2;
-    curlen = WEBRTC_SPL_RSHIFT_W16(frame_size, 1);
-
-    WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
-                  &inst->lower_state[filtno], curlen);
-
-    // Energy in 3000 Hz - 4000 Hz
-    curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
-    WebRtcVad_LogOfEnergy(vecHP2, &out_vector[5], &power, kOffsetVector[5], curlen);
-
-    // Energy in 2000 Hz - 3000 Hz
-    WebRtcVad_LogOfEnergy(vecLP2, &out_vector[4], &power, kOffsetVector[4], curlen);
-
-    // Split at 1000 Hz and downsample
-    filtno = 2;
-    ptin = vecLP1;
-    hptout = vecHP2;
-    lptout = vecLP2;
-    curlen = WEBRTC_SPL_RSHIFT_W16(frame_size, 1);
-    WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
-                  &inst->lower_state[filtno], curlen);
-
-    // Energy in 1000 Hz - 2000 Hz
-    curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
-    WebRtcVad_LogOfEnergy(vecHP2, &out_vector[3], &power, kOffsetVector[3], curlen);
-
-    // Split at 500 Hz
-    filtno = 3;
-    ptin = vecLP2;
-    hptout = vecHP1;
-    lptout = vecLP1;
-
-    WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
-                  &inst->lower_state[filtno], curlen);
-
-    // Energy in 500 Hz - 1000 Hz
-    curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
-    WebRtcVad_LogOfEnergy(vecHP1, &out_vector[2], &power, kOffsetVector[2], curlen);
-    // Split at 250 Hz
-    filtno = 4;
-    ptin = vecLP1;
-    hptout = vecHP2;
-    lptout = vecLP2;
-
-    WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
-                  &inst->lower_state[filtno], curlen);
-
-    // Energy in 250 Hz - 500 Hz
-    curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
-    WebRtcVad_LogOfEnergy(vecHP2, &out_vector[1], &power, kOffsetVector[1], curlen);
-
-    // Remove DC and LFs
-    WebRtcVad_HpOutput(vecLP2, curlen, vecHP1, inst->hp_filter_state);
-
-    // Power in 80 Hz - 250 Hz
-    WebRtcVad_LogOfEnergy(vecHP1, &out_vector[0], &power, kOffsetVector[0], curlen);
-
-    return power;
-}
-
-void WebRtcVad_LogOfEnergy(WebRtc_Word16 *vector,
-                           WebRtc_Word16 *enerlogval,
-                           WebRtc_Word16 *power,
-                           WebRtc_Word16 offset,
-                           int vector_length)
-{
-    WebRtc_Word16 enerSum = 0;
-    WebRtc_Word16 zeros, frac, log2;
-    WebRtc_Word32 energy;
-
-    int shfts = 0, shfts2;
-
-    energy = WebRtcSpl_Energy(vector, vector_length, &shfts);
-
-    if (energy > 0)
-    {
-
-        shfts2 = 16 - WebRtcSpl_NormW32(energy);
-        shfts += shfts2;
-        // "shfts" is the total number of right shifts that has been done to enerSum.
-        enerSum = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(energy, -shfts2);
-
-        // Find:
-        // 160*log10(enerSum*2^shfts) = 160*log10(2)*log2(enerSum*2^shfts) =
-        // 160*log10(2)*(log2(enerSum) + log2(2^shfts)) =
-        // 160*log10(2)*(log2(enerSum) + shfts)
-
-        zeros = WebRtcSpl_NormU32(enerSum);
-        frac = (WebRtc_Word16)(((WebRtc_UWord32)((WebRtc_Word32)(enerSum) << zeros)
-                & 0x7FFFFFFF) >> 21);
-        log2 = (WebRtc_Word16)(((31 - zeros) << 10) + frac);
-
-        *enerlogval = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(kLogConst, log2, 19)
-                + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(shfts, kLogConst, 9);
-
-        if (*enerlogval < 0)
-        {
-            *enerlogval = 0;
-        }
-    } else
-    {
-        *enerlogval = 0;
-        shfts = -15;
-        enerSum = 0;
-    }
-
-    *enerlogval += offset;
-
-    // Total power in frame
-    if (*power <= MIN_ENERGY)
-    {
-        if (shfts > 0)
-        {
-            *power += MIN_ENERGY + 1;
-        } else if (WEBRTC_SPL_SHIFT_W16(enerSum, shfts) > MIN_ENERGY)
-        {
-            *power += MIN_ENERGY + 1;
-        } else
-        {
-            *power += WEBRTC_SPL_SHIFT_W16(enerSum, shfts);
-        }
-    }
-}

src/common_audio/vad/main/source/vad_filterbank.h

@@ -1,143 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the description of the internal VAD call
- * WebRtcVad_GaussianProbability.
- */
-
-#ifndef WEBRTC_VAD_FILTERBANK_H_
-#define WEBRTC_VAD_FILTERBANK_H_
-
-#include "vad_core.h"
-
-/****************************************************************************
- * WebRtcVad_HpOutput(...)
- *
- * This function removes DC from the lowest frequency band
- *
- * Input:
- *      - in_vector         : Samples in the frequency interval 0 - 250 Hz
- *      - in_vector_length  : Length of input and output vector
- *      - filter_state      : Current state of the filter
- *
- * Output:
- *      - out_vector        : Samples in the frequency interval 80 - 250 Hz
- *      - filter_state      : Updated state of the filter
- *
- */
-void WebRtcVad_HpOutput(WebRtc_Word16* in_vector,
-                        WebRtc_Word16  in_vector_length,
-                        WebRtc_Word16* out_vector,
-                        WebRtc_Word16* filter_state);
-
-/****************************************************************************
- * WebRtcVad_Allpass(...)
- *
- * This function is used when before splitting a speech file into 
- * different frequency bands
- *
- * Note! Do NOT let the arrays in_vector and out_vector correspond to the same address.
- *
- * Input:
- *      - in_vector             : (Q0)
- *      - filter_coefficients   : (Q15)
- *      - vector_length         : Length of input and output vector
- *      - filter_state          : Current state of the filter (Q(-1))
- *
- * Output:
- *      - out_vector            : Output speech signal (Q(-1))
- *      - filter_state          : Updated state of the filter (Q(-1))
- *
- */
-void WebRtcVad_Allpass(WebRtc_Word16* in_vector,
-                       WebRtc_Word16* outw16,
-                       WebRtc_Word16 filter_coefficients,
-                       int vector_length,
-                       WebRtc_Word16* filter_state);
-
-/****************************************************************************
- * WebRtcVad_SplitFilter(...)
- *
- * This function is used when before splitting a speech file into 
- * different frequency bands
- *
- * Input:
- *      - in_vector         : Input signal to be split into two frequency bands.
- *      - upper_state       : Current state of the upper filter
- *      - lower_state       : Current state of the lower filter
- *      - in_vector_length  : Length of input vector
- *
- * Output:
- *      - out_vector_hp     : Upper half of the spectrum
- *      - out_vector_lp     : Lower half of the spectrum
- *      - upper_state       : Updated state of the upper filter
- *      - lower_state       : Updated state of the lower filter
- *
- */
-void WebRtcVad_SplitFilter(WebRtc_Word16* in_vector,
-                           WebRtc_Word16* out_vector_hp,
-                           WebRtc_Word16* out_vector_lp,
-                           WebRtc_Word16* upper_state,
-                           WebRtc_Word16* lower_state,
-                           int in_vector_length);
-
-/****************************************************************************
- * WebRtcVad_get_features(...)
- *
- * This function is used to get the logarithm of the power of each of the 
- * 6 frequency bands used by the VAD:
- *        80 Hz - 250 Hz
- *        250 Hz - 500 Hz
- *        500 Hz - 1000 Hz
- *        1000 Hz - 2000 Hz
- *        2000 Hz - 3000 Hz
- *        3000 Hz - 4000 Hz 
- *
- * Input:
- *      - inst        : Pointer to VAD instance
- *      - in_vector   : Input speech signal
- *      - frame_size  : Frame size, in number of samples
- *
- * Output:
- *      - out_vector  : 10*log10(power in each freq. band), Q4
- *    
- * Return: total power in the signal (NOTE! This value is not exact since it
- *         is only used in a comparison.
- */
-WebRtc_Word16 WebRtcVad_get_features(VadInstT* inst,
-                                     WebRtc_Word16* in_vector,
-                                     int frame_size,
-                                     WebRtc_Word16* out_vector);
-
-/****************************************************************************
- * WebRtcVad_LogOfEnergy(...)
- *
- * This function is used to get the logarithm of the power of one frequency band.
- *
- * Input:
- *      - vector            : Input speech samples for one frequency band
- *      - offset            : Offset value for the current frequency band
- *      - vector_length     : Length of input vector
- *
- * Output:
- *      - enerlogval        : 10*log10(energy);
- *      - power             : Update total power in speech frame. NOTE! This value
- *                            is not exact since it is only used in a comparison.
- *     
- */
-void WebRtcVad_LogOfEnergy(WebRtc_Word16* vector,
-                           WebRtc_Word16* enerlogval,
-                           WebRtc_Word16* power,
-                           WebRtc_Word16 offset,
-                           int vector_length);
-
-#endif // WEBRTC_VAD_FILTERBANK_H_

src/common_audio/vad/main/source/vad_gmm.c

@@ -1,75 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the internal VAD call
- * WebRtcVad_GaussianProbability. For function description, see vad_gmm.h.
- */
-
-#include "vad_gmm.h"
-
-#include "signal_processing_library.h"
-#include "typedefs.h"
-
-static const WebRtc_Word32 kCompVar = 22005;
-// Constant log2(exp(1)) in Q12
-static const WebRtc_Word16 kLog10Const = 5909;
-
-WebRtc_Word32 WebRtcVad_GaussianProbability(WebRtc_Word16 in_sample,
-                                            WebRtc_Word16 mean,
-                                            WebRtc_Word16 std,
-                                            WebRtc_Word16 *delta)
-{
-    WebRtc_Word16 tmp16, tmpDiv, tmpDiv2, expVal, tmp16_1, tmp16_2;
-    WebRtc_Word32 tmp32, y32;
-
-    // Calculate tmpDiv=1/std, in Q10
-    tmp32 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W16(std,1) + (WebRtc_Word32)131072; // 1 in Q17
-    tmpDiv = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32, std); // Q17/Q7 = Q10
-
-    // Calculate tmpDiv2=1/std^2, in Q14
-    tmp16 = WEBRTC_SPL_RSHIFT_W16(tmpDiv, 2); // From Q10 to Q8
-    tmpDiv2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 2); // (Q8 * Q8)>>2 = Q14
-
-    tmp16 = WEBRTC_SPL_LSHIFT_W16(in_sample, 3); // Q7
-    tmp16 = tmp16 - mean; // Q7 - Q7 = Q7
-
-    // To be used later, when updating noise/speech model
-    // delta = (x-m)/std^2, in Q11
-    *delta = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmpDiv2, tmp16, 10); //(Q14*Q7)>>10 = Q11
-
-    // Calculate tmp32=(x-m)^2/(2*std^2), in Q10
-    tmp32 = (WebRtc_Word32)WEBRTC_SPL_MUL_16_16_RSFT(*delta, tmp16, 9); // One shift for /2
-
-    // Calculate expVal ~= exp(-(x-m)^2/(2*std^2)) ~= exp2(-log2(exp(1))*tmp32)
-    if (tmp32 < kCompVar)
-    {
-        // Calculate tmp16 = log2(exp(1))*tmp32 , in Q10
-        tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)tmp32,
-                                                         kLog10Const, 12);
-        tmp16 = -tmp16;
-        tmp16_2 = (WebRtc_Word16)(0x0400 | (tmp16 & 0x03FF));
-        tmp16_1 = (WebRtc_Word16)(tmp16 ^ 0xFFFF);
-        tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(tmp16_1, 10);
-        tmp16 += 1;
-        // Calculate expVal=log2(-tmp32), in Q10
-        expVal = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)tmp16_2, tmp16);
-
-    } else
-    {
-        expVal = 0;
-    }
-
-    // Calculate y32=(1/std)*exp(-(x-m)^2/(2*std^2)), in Q20
-    y32 = WEBRTC_SPL_MUL_16_16(tmpDiv, expVal); // Q10 * Q10 = Q20
-
-    return y32; // Q20
-}

src/common_audio/vad/main/source/vad_gmm.h

@@ -1,47 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the description of the internal VAD call
- * WebRtcVad_GaussianProbability.
- */
-
-#ifndef WEBRTC_VAD_GMM_H_
-#define WEBRTC_VAD_GMM_H_
-
-#include "typedefs.h"
-
-/****************************************************************************
- * WebRtcVad_GaussianProbability(...)
- *
- * This function calculates the probability for the value 'in_sample', given that in_sample
- * comes from a normal distribution with mean 'mean' and standard deviation 'std'.
- *
- * Input:
- *      - in_sample     : Input sample in Q4
- *      - mean          : mean value in the statistical model, Q7
- *      - std           : standard deviation, Q7
- *
- * Output:
- *
- *      - delta         : Value used when updating the model, Q11
- *
- * Return:
- *      - out           : out = 1/std * exp(-(x-m)^2/(2*std^2));
- *                        Probability for x.
- *
- */
-WebRtc_Word32 WebRtcVad_GaussianProbability(WebRtc_Word16 in_sample,
-                                            WebRtc_Word16 mean,
-                                            WebRtc_Word16 std,
-                                            WebRtc_Word16 *delta);
-
-#endif // WEBRTC_VAD_GMM_H_

src/common_audio/vad/main/source/vad_sp.c

@@ -1,236 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the VAD internal calls for
- * Downsampling and FindMinimum.
- * For function call descriptions; See vad_sp.h.
- */
-
-#include "vad_sp.h"
-
-#include "signal_processing_library.h"
-#include "typedefs.h"
-#include "vad_defines.h"
-
-// Allpass filter coefficients, upper and lower, in Q13
-// Upper: 0.64, Lower: 0.17
-static const WebRtc_Word16 kAllPassCoefsQ13[2] = {5243, 1392}; // Q13
-
-// Downsampling filter based on the splitting filter and the allpass functions
-// in vad_filterbank.c
-void WebRtcVad_Downsampling(WebRtc_Word16* signal_in,
-                            WebRtc_Word16* signal_out,
-                            WebRtc_Word32* filter_state,
-                            int inlen)
-{
-    WebRtc_Word16 tmp16_1, tmp16_2;
-    WebRtc_Word32 tmp32_1, tmp32_2;
-    int n, halflen;
-
-    // Downsampling by 2 and get two branches
-    halflen = WEBRTC_SPL_RSHIFT_W16(inlen, 1);
-
-    tmp32_1 = filter_state[0];
-    tmp32_2 = filter_state[1];
-
-    // Filter coefficients in Q13, filter state in Q0
-    for (n = 0; n < halflen; n++)
-    {
-        // All-pass filtering upper branch
-        tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32_1, 1)
-                + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[0]),
-                                                           *signal_in, 14);
-        *signal_out = tmp16_1;
-        tmp32_1 = (WebRtc_Word32)(*signal_in++)
-                - (WebRtc_Word32)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[0]), tmp16_1, 12);
-
-        // All-pass filtering lower branch
-        tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32_2, 1)
-                + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[1]),
-                                                           *signal_in, 14);
-        *signal_out++ += tmp16_2;
-        tmp32_2 = (WebRtc_Word32)(*signal_in++)
-                - (WebRtc_Word32)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[1]), tmp16_2, 12);
-    }
-    filter_state[0] = tmp32_1;
-    filter_state[1] = tmp32_2;
-}
-
-WebRtc_Word16 WebRtcVad_FindMinimum(VadInstT* inst,
-                                    WebRtc_Word16 x,
-                                    int n)
-{
-    int i, j, k, II = -1, offset;
-    WebRtc_Word16 meanV, alpha;
-    WebRtc_Word32 tmp32, tmp32_1;
-    WebRtc_Word16 *valptr, *idxptr, *p1, *p2, *p3;
-
-    // Offset to beginning of the 16 minimum values in memory
-    offset = WEBRTC_SPL_LSHIFT_W16(n, 4);
-
-    // Pointer to memory for the 16 minimum values and the age of each value
-    idxptr = &inst->index_vector[offset];
-    valptr = &inst->low_value_vector[offset];
-
-    // Each value in low_value_vector is getting 1 loop older.
-    // Update age of each value in indexVal, and remove old values.
-    for (i = 0; i < 16; i++)
-    {
-        p3 = idxptr + i;
-        if (*p3 != 100)
-        {
-            *p3 += 1;
-        } else
-        {
-            p1 = valptr + i + 1;
-            p2 = p3 + 1;
-            for (j = i; j < 16; j++)
-            {
-                *(valptr + j) = *p1++;
-                *(idxptr + j) = *p2++;
-            }
-            *(idxptr + 15) = 101;
-            *(valptr + 15) = 10000;
-        }
-    }
-
-    // Check if x smaller than any of the values in low_value_vector.
-    // If so, find position.
-    if (x < *(valptr + 7))
-    {
-        if (x < *(valptr + 3))
-        {
-            if (x < *(valptr + 1))
-            {
-                if (x < *valptr)
-                {
-                    II = 0;
-                } else
-                {
-                    II = 1;
-                }
-            } else if (x < *(valptr + 2))
-            {
-                II = 2;
-            } else
-            {
-                II = 3;
-            }
-        } else if (x < *(valptr + 5))
-        {
-            if (x < *(valptr + 4))
-            {
-                II = 4;
-            } else
-            {
-                II = 5;
-            }
-        } else if (x < *(valptr + 6))
-        {
-            II = 6;
-        } else
-        {
-            II = 7;
-        }
-    } else if (x < *(valptr + 15))
-    {
-        if (x < *(valptr + 11))
-        {
-            if (x < *(valptr + 9))
-            {
-                if (x < *(valptr + 8))
-                {
-                    II = 8;
-                } else
-                {
-                    II = 9;
-                }
-            } else if (x < *(valptr + 10))
-            {
-                II = 10;
-            } else
-            {
-                II = 11;
-            }
-        } else if (x < *(valptr + 13))
-        {
-            if (x < *(valptr + 12))
-            {
-                II = 12;
-            } else
-            {
-                II = 13;
-            }
-        } else if (x < *(valptr + 14))
-        {
-            II = 14;
-        } else
-        {
-            II = 15;
-        }
-    }
-
-    // Put new min value on right position and shift bigger values up
-    if (II > -1)
-    {
-        for (i = 15; i > II; i--)
-        {
-            k = i - 1;
-            *(valptr + i) = *(valptr + k);
-            *(idxptr + i) = *(idxptr + k);
-        }
-        *(valptr + II) = x;
-        *(idxptr + II) = 1;
-    }
-
-    meanV = 0;
-    if ((inst->frame_counter) > 4)
-    {
-        j = 5;
-    } else
-    {
-        j = inst->frame_counter;
-    }
-
-    if (j > 2)
-    {
-        meanV = *(valptr + 2);
-    } else if (j > 0)
-    {
-        meanV = *valptr;
-    } else
-    {
-        meanV = 1600;
-    }
-
-    if (inst->frame_counter > 0)
-    {
-        if (meanV < inst->mean_value[n])
-        {
-            alpha = (WebRtc_Word16)ALPHA1; // 0.2 in Q15
-        } else
-        {
-            alpha = (WebRtc_Word16)ALPHA2; // 0.99 in Q15
-        }
-    } else
-    {
-        alpha = 0;
-    }
-
-    tmp32 = WEBRTC_SPL_MUL_16_16((alpha+1), inst->mean_value[n]);
-    tmp32_1 = WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_WORD16_MAX - alpha, meanV);
-    tmp32 += tmp32_1;
-    tmp32 += 16384;
-    inst->mean_value[n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 15);
-
-    return inst->mean_value[n];
-}

src/common_audio/vad/main/source/vad_sp.h

@@ -1,60 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the VAD internal calls for Downsampling and FindMinimum.
- * Specific function calls are given below.
- */
-
-#ifndef WEBRTC_VAD_SP_H_
-#define WEBRTC_VAD_SP_H_
-
-#include "vad_core.h"
-
-/****************************************************************************
- * WebRtcVad_Downsampling(...)
- *
- * Downsamples the signal a factor 2, eg. 32->16 or 16->8
- *
- * Input:
- *      - signal_in     : Input signal
- *      - in_length     : Length of input signal in samples
- *
- * Input & Output:
- *      - filter_state  : Filter state for first all-pass filters
- *
- * Output:
- *      - signal_out    : Downsampled signal (of length len/2)
- */
-void WebRtcVad_Downsampling(WebRtc_Word16* signal_in,
-                            WebRtc_Word16* signal_out,
-                            WebRtc_Word32* filter_state,
-                            int in_length);
-
-/****************************************************************************
- * WebRtcVad_FindMinimum(...)
- *
- * Find the five lowest values of x in 100 frames long window. Return a mean
- * value of these five values.
- *
- * Input:
- *      - feature_value : Feature value
- *      - channel       : Channel number
- *
- * Input & Output:
- *      - inst          : State information
- *
- * Output:
- *      return value    : Weighted minimum value for a moving window.
- */
-WebRtc_Word16 WebRtcVad_FindMinimum(VadInstT* inst, WebRtc_Word16 feature_value, int channel);
-
-#endif // WEBRTC_VAD_SP_H_

src/common_audio/vad/main/source/webrtc_vad.c

@@ -1,197 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the VAD API calls. For a specific function call description,
- * see webrtc_vad.h
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "webrtc_vad.h"
-#include "vad_core.h"
-
-static const int kInitCheck = 42;
-
-WebRtc_Word16 WebRtcVad_get_version(char *version, size_t size_bytes)
-{
-    const char my_version[] = "VAD 1.2.0";
-
-    if (version == NULL)
-    {
-        return -1;
-    }
-
-    if (size_bytes < sizeof(my_version))
-    {
-        return -1;
-    }
-
-    memcpy(version, my_version, sizeof(my_version));
-    return 0;
-}
-
-WebRtc_Word16 WebRtcVad_AssignSize(int *size_in_bytes)
-{
-    *size_in_bytes = sizeof(VadInstT) * 2 / sizeof(WebRtc_Word16);
-    return 0;
-}
-
-WebRtc_Word16 WebRtcVad_Assign(VadInst **vad_inst, void *vad_inst_addr)
-{
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    if (vad_inst_addr != NULL)
-    {
-        *vad_inst = (VadInst*)vad_inst_addr;
-        return 0;
-    } else
-    {
-        return -1;
-    }
-}
-
-WebRtc_Word16 WebRtcVad_Create(VadInst **vad_inst)
-{
-
-    VadInstT *vad_ptr = NULL;
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    *vad_inst = NULL;
-
-    vad_ptr = (VadInstT *)malloc(sizeof(VadInstT));
-    *vad_inst = (VadInst *)vad_ptr;
-
-    if (vad_ptr == NULL)
-    {
-        return -1;
-    }
-
-    vad_ptr->init_flag = 0;
-
-    return 0;
-}
-
-WebRtc_Word16 WebRtcVad_Free(VadInst *vad_inst)
-{
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    free(vad_inst);
-    return 0;
-}
-
-WebRtc_Word16 WebRtcVad_Init(VadInst *vad_inst)
-{
-    short mode = 0; // Default high quality
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    return WebRtcVad_InitCore((VadInstT*)vad_inst, mode);
-}
-
-WebRtc_Word16 WebRtcVad_set_mode(VadInst *vad_inst, WebRtc_Word16 mode)
-{
-    VadInstT* vad_ptr;
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    vad_ptr = (VadInstT*)vad_inst;
-    if (vad_ptr->init_flag != kInitCheck)
-    {
-        return -1;
-    }
-
-    return WebRtcVad_set_mode_core((VadInstT*)vad_inst, mode);
-}
-
-WebRtc_Word16 WebRtcVad_Process(VadInst *vad_inst,
-                                WebRtc_Word16 fs,
-                                WebRtc_Word16 *speech_frame,
-                                WebRtc_Word16 frame_length)
-{
-    WebRtc_Word16 vad;
-    VadInstT* vad_ptr;
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    vad_ptr = (VadInstT*)vad_inst;
-    if (vad_ptr->init_flag != kInitCheck)
-    {
-        return -1;
-    }
-
-    if (speech_frame == NULL)
-    {
-        return -1;
-    }
-
-    if (fs == 32000)
-    {
-        if ((frame_length != 320) && (frame_length != 640) && (frame_length != 960))
-        {
-            return -1;
-        }
-        vad = WebRtcVad_CalcVad32khz((VadInstT*)vad_inst, speech_frame, frame_length);
-
-    } else if (fs == 16000)
-    {
-        if ((frame_length != 160) && (frame_length != 320) && (frame_length != 480))
-        {
-            return -1;
-        }
-        vad = WebRtcVad_CalcVad16khz((VadInstT*)vad_inst, speech_frame, frame_length);
-
-    } else if (fs == 8000)
-    {
-        if ((frame_length != 80) && (frame_length != 160) && (frame_length != 240))
-        {
-            return -1;
-        }
-        vad = WebRtcVad_CalcVad8khz((VadInstT*)vad_inst, speech_frame, frame_length);
-
-    } else
-    {
-        return -1; // Not a supported sampling frequency
-    }
-
-    if (vad > 0)
-    {
-        return 1;
-    } else if (vad == 0)
-    {
-        return 0;
-    } else
-    {
-        return -1;
-    }
-}

src/common_audio/vad/main/test/unit_test/unit_test.cc

@@ -1,123 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the VAD unit tests.
- */
-
-#include <cstring>
-#include "unit_test.h"
-#include "webrtc_vad.h"
-
-
-class VadEnvironment : public ::testing::Environment {
- public:
-  virtual void SetUp() {
-  }
-
-  virtual void TearDown() {
-  }
-};
-
-VadTest::VadTest()
-{
-}
-
-void VadTest::SetUp() {
-}
-
-void VadTest::TearDown() {
-}
-
-TEST_F(VadTest, ApiTest) {
-    VadInst *vad_inst;
-    int i, j, k;
-    short zeros[960];
-    short speech[960];
-    char version[32];
-
-    // Valid test cases
-    int fs[3] = {8000, 16000, 32000};
-    int nMode[4] = {0, 1, 2, 3};
-    int framelen[3][3] = {{80, 160, 240},
-    {160, 320, 480}, {320, 640, 960}} ;
-    int vad_counter = 0;
-
-    memset(zeros, 0, sizeof(short) * 960);
-    memset(speech, 1, sizeof(short) * 960);
-    speech[13] = 1374;
-    speech[73] = -3747;
-
-
-
-    // WebRtcVad_get_version()
-    WebRtcVad_get_version(version);
-    //printf("API Test for %s\n", version);
-
-    // Null instance tests
-    EXPECT_EQ(-1, WebRtcVad_Create(NULL));
-    EXPECT_EQ(-1, WebRtcVad_Init(NULL));
-    EXPECT_EQ(-1, WebRtcVad_Assign(NULL, NULL));
-    EXPECT_EQ(-1, WebRtcVad_Free(NULL));
-    EXPECT_EQ(-1, WebRtcVad_set_mode(NULL, nMode[0]));
-    EXPECT_EQ(-1, WebRtcVad_Process(NULL, fs[0], speech,  framelen[0][0]));
-
-
-    EXPECT_EQ(WebRtcVad_Create(&vad_inst), 0);
-
-    // Not initialized tests
-    EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, fs[0], speech,  framelen[0][0]));
-    EXPECT_EQ(-1, WebRtcVad_set_mode(vad_inst, nMode[0]));
-
-    // WebRtcVad_Init() tests
-    EXPECT_EQ(WebRtcVad_Init(vad_inst), 0);
-
-    // WebRtcVad_set_mode() tests
-    EXPECT_EQ(-1, WebRtcVad_set_mode(vad_inst, -1));
-    EXPECT_EQ(-1, WebRtcVad_set_mode(vad_inst, 4));
-
-    for (i = 0; i < sizeof(nMode)/sizeof(nMode[0]); i++) {
-        EXPECT_EQ(WebRtcVad_set_mode(vad_inst, nMode[i]), 0);
-    }
-
-    // WebRtcVad_Process() tests
-    EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, fs[0], NULL,  framelen[0][0]));
-    EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, 12000, speech,  framelen[0][0]));
-    EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, fs[0], speech,  framelen[1][1]));
-    EXPECT_EQ(WebRtcVad_Process(vad_inst, fs[0], zeros,  framelen[0][0]), 0);
-    for (i = 0; i < sizeof(fs)/sizeof(fs[0]); i++) {
-        for (j = 0; j < sizeof(framelen[0])/sizeof(framelen[0][0]); j++) {
-            for (k = 0; k < sizeof(nMode)/sizeof(nMode[0]); k++) {
-                EXPECT_EQ(WebRtcVad_set_mode(vad_inst, nMode[k]), 0);
-//                printf("%d\n", WebRtcVad_Process(vad_inst, fs[i], speech,  framelen[i][j]));
-                if (vad_counter < 9)
-                {
-                    EXPECT_EQ(WebRtcVad_Process(vad_inst, fs[i], speech,  framelen[i][j]), 1);
-                } else
-                {
-                    EXPECT_EQ(WebRtcVad_Process(vad_inst, fs[i], speech,  framelen[i][j]), 0);
-                }
-                vad_counter++;
-            }
-        }
-    }
-
-    EXPECT_EQ(0, WebRtcVad_Free(vad_inst));
-
-}
-
-int main(int argc, char** argv) {
-  ::testing::InitGoogleTest(&argc, argv);
-  VadEnvironment* env = new VadEnvironment;
-  ::testing::AddGlobalTestEnvironment(env);
-
-  return RUN_ALL_TESTS();
-}

src/common_audio/vad/main/test/unit_test/unit_test.h

@@ -1,28 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the declaration of the VAD unit test.
- */
-
-#ifndef WEBRTC_VAD_UNIT_TEST_H_
-#define WEBRTC_VAD_UNIT_TEST_H_
-
-#include <gtest/gtest.h>
-
-class VadTest : public ::testing::Test {
- protected:
-  VadTest();
-  virtual void SetUp();
-  virtual void TearDown();
-};
-
-#endif  // WEBRTC_VAD_UNIT_TEST_H_

src/common_types.h

@@ -1,610 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_COMMON_TYPES_H
-#define WEBRTC_COMMON_TYPES_H
-
-#include "typedefs.h"
-
-#ifdef WEBRTC_EXPORT
-    #define WEBRTC_DLLEXPORT _declspec(dllexport)
-#elif WEBRTC_DLL
-    #define WEBRTC_DLLEXPORT _declspec(dllimport)
-#else
-    #define WEBRTC_DLLEXPORT
-#endif
-
-#ifndef NULL
-    #define NULL 0
-#endif
-
-namespace webrtc {
-
-class InStream
-{
-public:
-    virtual int Read(void *buf,int len) = 0;
-    virtual int Rewind() {return -1;}
-    virtual ~InStream() {}
-protected:
-    InStream() {}
-};
-
-class OutStream
-{
-public:
-    virtual bool Write(const void *buf,int len) = 0;
-    virtual int Rewind() {return -1;}
-    virtual ~OutStream() {}
-protected:
-    OutStream() {}
-};
-
-enum TraceModule
-{
-    // not a module, triggered from the engine code
-    kTraceVoice              = 0x0001,
-    // not a module, triggered from the engine code
-    kTraceVideo              = 0x0002,
-    // not a module, triggered from the utility code
-    kTraceUtility            = 0x0003,
-    kTraceRtpRtcp            = 0x0004,
-    kTraceTransport          = 0x0005,
-    kTraceSrtp               = 0x0006,
-    kTraceAudioCoding        = 0x0007,
-    kTraceAudioMixerServer   = 0x0008,
-    kTraceAudioMixerClient   = 0x0009,
-    kTraceFile               = 0x000a,
-    kTraceAudioProcessing    = 0x000b,
-    kTraceVideoCoding        = 0x0010,
-    kTraceVideoMixer         = 0x0011,
-    kTraceAudioDevice        = 0x0012,
-    kTraceVideoRenderer      = 0x0014,
-    kTraceVideoCapture       = 0x0015,
-    kTraceVideoPreocessing   = 0x0016
-};
-
-enum TraceLevel
-{
-    kTraceNone               = 0x0000,    // no trace
-    kTraceStateInfo          = 0x0001,
-    kTraceWarning            = 0x0002,
-    kTraceError              = 0x0004,
-    kTraceCritical           = 0x0008,
-    kTraceApiCall            = 0x0010,
-    kTraceDefault            = 0x00ff,
-
-    kTraceModuleCall         = 0x0020,
-    kTraceMemory             = 0x0100,   // memory info
-    kTraceTimer              = 0x0200,   // timing info
-    kTraceStream             = 0x0400,   // "continuous" stream of data
-
-    // used for debug purposes
-    kTraceDebug              = 0x0800,  // debug
-    kTraceInfo               = 0x1000,  // debug info
-
-    kTraceAll                = 0xffff
-};
-
-// External Trace API
-class TraceCallback
-{
-public:
-    virtual void Print(const TraceLevel level,
-                       const char *traceString,
-                       const int length) = 0;
-protected:
-    virtual ~TraceCallback() {}
-    TraceCallback() {}
-};
-
-
-enum FileFormats
-{
-    kFileFormatWavFile        = 1,
-    kFileFormatCompressedFile = 2,
-    kFileFormatAviFile        = 3,
-    kFileFormatPreencodedFile = 4,
-    kFileFormatPcm16kHzFile   = 7,
-    kFileFormatPcm8kHzFile    = 8,
-    kFileFormatPcm32kHzFile   = 9
-};
-
-
-enum ProcessingTypes
-{
-    kPlaybackPerChannel = 0,
-    kPlaybackAllChannelsMixed,
-    kRecordingPerChannel,
-    kRecordingAllChannelsMixed
-};
-
-// Encryption enums
-enum CipherTypes
-{
-    kCipherNull               = 0,
-    kCipherAes128CounterMode  = 1
-};
-
-enum AuthenticationTypes
-{
-    kAuthNull       = 0,
-    kAuthHmacSha1   = 3
-};
-
-enum SecurityLevels
-{
-    kNoProtection                    = 0,
-    kEncryption                      = 1,
-    kAuthentication                  = 2,
-    kEncryptionAndAuthentication     = 3
-};
-
-class Encryption
-{
-public:
-    virtual void encrypt(
-        int channel_no,
-        unsigned char* in_data,
-        unsigned char* out_data,
-        int bytes_in,
-        int* bytes_out) = 0;
-
-    virtual void decrypt(
-        int channel_no,
-        unsigned char* in_data,
-        unsigned char* out_data,
-        int bytes_in,
-        int* bytes_out) = 0;
-
-    virtual void encrypt_rtcp(
-        int channel_no,
-        unsigned char* in_data,
-        unsigned char* out_data,
-        int bytes_in,
-        int* bytes_out) = 0;
-
-    virtual void decrypt_rtcp(
-        int channel_no,
-        unsigned char* in_data,
-        unsigned char* out_data,
-        int bytes_in,
-        int* bytes_out) = 0;
-
-protected:
-    virtual ~Encryption() {}
-    Encryption() {}
-};
-
-// External transport callback interface
-class Transport
-{
-public:
-    virtual int SendPacket(int channel, const void *data, int len) = 0;
-    virtual int SendRTCPPacket(int channel, const void *data, int len) = 0;
-
-protected:
-    virtual ~Transport() {}
-    Transport() {}
-};
-
-// ==================================================================
-// Voice specific types
-// ==================================================================
-
-// Each codec supported can be described by this structure.
-struct CodecInst
-{
-    int pltype;
-    char plname[32];
-    int plfreq;
-    int pacsize;
-    int channels;
-    int rate;
-};
-
-enum FrameType
-{
-    kFrameEmpty            = 0,
-    kAudioFrameSpeech      = 1,
-    kAudioFrameCN          = 2,
-    kVideoFrameKey         = 3,    // independent frame
-    kVideoFrameDelta       = 4,    // depends on the previus frame
-    kVideoFrameGolden      = 5,    // depends on a old known previus frame
-    kVideoFrameAltRef      = 6
-};
-
-// RTP
-enum {kRtpCsrcSize = 15}; // RFC 3550 page 13
-
-enum RTPDirections
-{
-    kRtpIncoming = 0,
-    kRtpOutgoing
-};
-
-enum PayloadFrequencies
-{
-    kFreq8000Hz = 8000,
-    kFreq16000Hz = 16000,
-    kFreq32000Hz = 32000
-};
-
-enum VadModes                 // degree of bandwidth reduction
-{
-    kVadConventional = 0,      // lowest reduction
-    kVadAggressiveLow,
-    kVadAggressiveMid,
-    kVadAggressiveHigh         // highest reduction
-};
-
-struct NetworkStatistics           // NETEQ statistics
-{
-    // current jitter buffer size in ms
-    WebRtc_UWord16 currentBufferSize;
-    // preferred (optimal) buffer size in ms
-    WebRtc_UWord16 preferredBufferSize;
-    // loss rate (network + late) in percent (in Q14)
-    WebRtc_UWord16 currentPacketLossRate;
-    // late loss rate in percent (in Q14)
-    WebRtc_UWord16 currentDiscardRate;
-    // fraction (of original stream) of synthesized speech inserted through
-    // expansion (in Q14)
-    WebRtc_UWord16 currentExpandRate;
-    // fraction of synthesized speech inserted through pre-emptive expansion
-    // (in Q14)
-    WebRtc_UWord16 currentPreemptiveRate;
-    // fraction of data removed through acceleration (in Q14)
-    WebRtc_UWord16 currentAccelerateRate;
-};
-
-struct JitterStatistics
-{
-    // smallest Jitter Buffer size during call in ms
-    WebRtc_UWord32 jbMinSize;
-    // largest Jitter Buffer size during call in ms
-    WebRtc_UWord32 jbMaxSize;
-    // the average JB size, measured over time - ms
-    WebRtc_UWord32 jbAvgSize;
-    // number of times the Jitter Buffer changed (using Accelerate or
-    // Pre-emptive Expand)
-    WebRtc_UWord32 jbChangeCount;
-    // amount (in ms) of audio data received late
-    WebRtc_UWord32 lateLossMs;
-    // milliseconds removed to reduce jitter buffer size
-    WebRtc_UWord32 accelerateMs;
-    // milliseconds discarded through buffer flushing
-    WebRtc_UWord32 flushedMs;
-    // milliseconds of generated silence
-    WebRtc_UWord32 generatedSilentMs;
-    // milliseconds of synthetic audio data (non-background noise)
-    WebRtc_UWord32 interpolatedVoiceMs;
-    // milliseconds of synthetic audio data (background noise level)
-    WebRtc_UWord32 interpolatedSilentMs;
-    // count of tiny expansions in output audio
-    WebRtc_UWord32 countExpandMoreThan120ms;
-    // count of small expansions in output audio
-    WebRtc_UWord32 countExpandMoreThan250ms;
-    // count of medium expansions in output audio
-    WebRtc_UWord32 countExpandMoreThan500ms;
-    // count of long expansions in output audio
-    WebRtc_UWord32 countExpandMoreThan2000ms;
-    // duration of longest audio drop-out
-    WebRtc_UWord32 longestExpandDurationMs;
-    // count of times we got small network outage (inter-arrival time in
-    // [500, 1000) ms)
-    WebRtc_UWord32 countIAT500ms;
-    // count of times we got medium network outage (inter-arrival time in
-    // [1000, 2000) ms)
-    WebRtc_UWord32 countIAT1000ms;
-    // count of times we got large network outage (inter-arrival time >=
-    // 2000 ms)
-    WebRtc_UWord32 countIAT2000ms;
-    // longest packet inter-arrival time in ms
-    WebRtc_UWord32 longestIATms;
-    // min time incoming Packet "waited" to be played
-    WebRtc_UWord32 minPacketDelayMs;
-    // max time incoming Packet "waited" to be played
-    WebRtc_UWord32 maxPacketDelayMs;
-    // avg time incoming Packet "waited" to be played
-    WebRtc_UWord32 avgPacketDelayMs;
-};
-
-typedef struct
-{
-    int min;              // minumum
-    int max;              // maximum
-    int average;          // average
-} StatVal;
-
-typedef struct           // All levels are reported in dBm0
-{
-    StatVal speech_rx;   // long-term speech levels on receiving side
-    StatVal speech_tx;   // long-term speech levels on transmitting side
-    StatVal noise_rx;    // long-term noise/silence levels on receiving side
-    StatVal noise_tx;    // long-term noise/silence levels on transmitting side
-} LevelStatistics;
-
-typedef struct        // All levels are reported in dB
-{
-    StatVal erl;      // Echo Return Loss
-    StatVal erle;     // Echo Return Loss Enhancement
-    StatVal rerl;     // RERL = ERL + ERLE
-    // Echo suppression inside EC at the point just before its NLP
-    StatVal a_nlp;
-} EchoStatistics;
-
-enum TelephoneEventDetectionMethods
-{
-    kInBand = 0,
-    kOutOfBand = 1,
-    kInAndOutOfBand = 2
-};
-
-enum NsModes    // type of Noise Suppression
-{
-    kNsUnchanged = 0,   // previously set mode
-    kNsDefault,         // platform default
-    kNsConference,      // conferencing default
-    kNsLowSuppression,  // lowest suppression
-    kNsModerateSuppression,
-    kNsHighSuppression,
-    kNsVeryHighSuppression,     // highest suppression
-};
-
-enum AgcModes                  // type of Automatic Gain Control
-{
-    kAgcUnchanged = 0,        // previously set mode
-    kAgcDefault,              // platform default
-    // adaptive mode for use when analog volume control exists (e.g. for
-    // PC softphone)
-    kAgcAdaptiveAnalog,
-    // scaling takes place in the digital domain (e.g. for conference servers
-    // and embedded devices)
-    kAgcAdaptiveDigital,
-    // can be used on embedded devices where the the capture signal is level
-    // is predictable
-    kAgcFixedDigital
-};
-
-// EC modes
-enum EcModes                   // type of Echo Control
-{
-    kEcUnchanged = 0,          // previously set mode
-    kEcDefault,                // platform default
-    kEcConference,             // conferencing default (aggressive AEC)
-    kEcAec,                    // Acoustic Echo Cancellation
-    kEcAecm,                   // AEC mobile
-};
-
-// AECM modes
-enum AecmModes                 // mode of AECM
-{
-    kAecmQuietEarpieceOrHeadset = 0,
-                               // Quiet earpiece or headset use
-    kAecmEarpiece,             // most earpiece use
-    kAecmLoudEarpiece,         // Loud earpiece or quiet speakerphone use
-    kAecmSpeakerphone,         // most speakerphone use (default)
-    kAecmLoudSpeakerphone      // Loud speakerphone
-};
-
-// AGC configuration
-typedef struct
-{
-    unsigned short targetLeveldBOv;
-    unsigned short digitalCompressionGaindB;
-    bool           limiterEnable;
-} AgcConfig;                  // AGC configuration parameters
-
-enum StereoChannel
-{
-    kStereoLeft = 0,
-    kStereoRight,
-    kStereoBoth
-};
-
-// Audio device layers
-enum AudioLayers
-{
-    kAudioPlatformDefault = 0,
-    kAudioWindowsWave = 1,
-    kAudioWindowsCore = 2,
-    kAudioLinuxAlsa = 3,
-    kAudioLinuxPulse = 4
-};
-
-enum NetEqModes             // NetEQ playout configurations
-{
-    // Optimized trade-off between low delay and jitter robustness for two-way
-    // communication.
-    kNetEqDefault = 0,
-    // Improved jitter robustness at the cost of increased delay. Can be
-    // used in one-way communication.
-    kNetEqStreaming = 1,
-    // Optimzed for decodability of fax signals rather than for perceived audio
-    // quality.
-    kNetEqFax = 2,
-};
-
-enum NetEqBgnModes          // NetEQ Background Noise (BGN) configurations
-{
-    // BGN is always on and will be generated when the incoming RTP stream
-    // stops (default).
-    kBgnOn = 0,
-    // The BGN is faded to zero (complete silence) after a few seconds.
-    kBgnFade = 1,
-    // BGN is not used at all. Silence is produced after speech extrapolation
-    // has faded.
-    kBgnOff = 2,
-};
-
-enum OnHoldModes            // On Hold direction
-{
-    kHoldSendAndPlay = 0,    // Put both sending and playing in on-hold state.
-    kHoldSendOnly,           // Put only sending in on-hold state.
-    kHoldPlayOnly            // Put only playing in on-hold state.
-};
-
-enum AmrMode
-{
-    kRfc3267BwEfficient = 0,
-    kRfc3267OctetAligned = 1,
-    kRfc3267FileStorage = 2,
-};
-
-// ==================================================================
-// Video specific types
-// ==================================================================
-
-// Raw video types
-enum RawVideoType
-{
-    kVideoI420     = 0,
-    kVideoYV12     = 1,
-    kVideoYUY2     = 2,
-    kVideoUYVY     = 3,
-    kVideoIYUV     = 4,
-    kVideoARGB     = 5,
-    kVideoRGB24    = 6,
-    kVideoRGB565   = 7,
-    kVideoARGB4444 = 8,
-    kVideoARGB1555 = 9,
-    kVideoMJPEG    = 10,
-    kVideoNV12     = 11,
-    kVideoNV21     = 12,
-    kVideoUnknown  = 99
-};
-
-// Video codec
-enum { kConfigParameterSize = 128};
-enum { kPayloadNameSize = 32};
-enum { kMaxSimulcastStreams = 4};
-
-// H.263 specific
-struct VideoCodecH263
-{
-    char quality;
-};
-
-// H.264 specific
-enum H264Packetization
-{
-    kH264SingleMode         = 0,
-    kH264NonInterleavedMode = 1
-};
-
-enum VideoCodecComplexity
-{
-    kComplexityNormal = 0,
-    kComplexityHigh    = 1,
-    kComplexityHigher  = 2,
-    kComplexityMax     = 3
-};
-
-enum VideoCodecProfile
-{
-    kProfileBase = 0x00,
-    kProfileMain = 0x01
-};
-
-struct VideoCodecH264
-{
-    H264Packetization          packetization;
-    VideoCodecComplexity       complexity;
-    VideoCodecProfile          profile;
-    char                       level;
-    char                       quality;
-
-    bool                       useFMO;
-
-    unsigned char              configParameters[kConfigParameterSize];
-    unsigned char              configParametersSize;
-};
-
-// VP8 specific
-struct VideoCodecVP8
-{
-    bool                 pictureLossIndicationOn;
-    bool                 feedbackModeOn;
-    VideoCodecComplexity complexity;
-    unsigned char        numberOfTemporalLayers;
-};
-
-// MPEG-4 specific
-struct VideoCodecMPEG4
-{
-    unsigned char   configParameters[kConfigParameterSize];
-    unsigned char   configParametersSize;
-    char            level;
-};
-
-// Unknown specific
-struct VideoCodecGeneric
-{
-};
-
-// Video codec types
-enum VideoCodecType
-{
-    kVideoCodecH263,
-    kVideoCodecH264,
-    kVideoCodecVP8,
-    kVideoCodecMPEG4,
-    kVideoCodecI420,
-    kVideoCodecRED,
-    kVideoCodecULPFEC,
-    kVideoCodecUnknown
-};
-
-union VideoCodecUnion
-{
-    VideoCodecH263      H263;
-    VideoCodecH264      H264;
-    VideoCodecVP8       VP8;
-    VideoCodecMPEG4     MPEG4;
-    VideoCodecGeneric   Generic;
-};
-
-/*
-*  Simulcast is when the same stream is encoded multiple times with different
-*  settings such as resolution.  
-*/
-struct SimulcastStream
-{
-    unsigned short      width;
-    unsigned short      height;
-    unsigned char       numberOfTemporalLayers;
-    unsigned int        maxBitrate;
-    unsigned int        qpMax; // minimum quality
-};
-
-// Common video codec properties
-struct VideoCodec
-{
-    VideoCodecType      codecType;
-    char                plName[kPayloadNameSize];
-    unsigned char       plType;
-
-    unsigned short      width;
-    unsigned short      height;
-
-    unsigned int        startBitrate;
-    unsigned int        maxBitrate;
-    unsigned int        minBitrate;
-    unsigned char       maxFramerate;
-
-    VideoCodecUnion     codecSpecific;
-
-    unsigned int        qpMax;
-    unsigned char       numberOfSimulcastStreams;
-    SimulcastStream     simulcastStream[kMaxSimulcastStreams];
-};
-}  // namespace webrtc
-#endif  // WEBRTC_COMMON_TYPES_H

src/modules/Makefile.am

@@ -1 +0,0 @@
-SUBDIRS = audio_processing

src/modules/audio_processing/Makefile.am

@@ -1,59 +0,0 @@
-SUBDIRS = utility ns aec aecm agc
-lib_LTLIBRARIES = libwebrtc_audio_processing.la
-
-if NS_FIXED
-COMMON_CXXFLAGS += -DWEBRTC_NS_FIXED=1
-NS_LIB = libns_fix
-else
-COMMON_CXXFLAGS += -DWEBRTC_NS_FLOAT=1
-NS_LIB = libns
-endif
-
-webrtcincludedir = $(includedir)/webrtc_audio_processing
-webrtcinclude_HEADERS = $(top_srcdir)/src/typedefs.h \
-			$(top_srcdir)/src/modules/interface/module.h \
-			interface/audio_processing.h \
-			$(top_srcdir)/src/common_types.h \
-			$(top_srcdir)/src/modules/interface/module_common_types.h
-
-libwebrtc_audio_processing_la_SOURCES = interface/audio_processing.h \
-					audio_buffer.cc \
-					audio_buffer.h \
-					audio_processing_impl.cc \
-					audio_processing_impl.h \
-					echo_cancellation_impl.cc \
-					echo_cancellation_impl.h \
-					echo_control_mobile_impl.cc \
-					echo_control_mobile_impl.h \
-					gain_control_impl.cc \
-					gain_control_impl.h \
-					high_pass_filter_impl.cc \
-					high_pass_filter_impl.h \
-					level_estimator_impl.cc \
-					level_estimator_impl.h \
-					noise_suppression_impl.cc \
-					noise_suppression_impl.h \
-					splitting_filter.cc \
-					splitting_filter.h \
-					processing_component.cc \
-					processing_component.h \
-					voice_detection_impl.cc \
-					voice_detection_impl.h
-libwebrtc_audio_processing_la_CXXFLAGS = $(AM_CXXFLAGS) $(COMMON_CXXFLAGS) \
-				       -I$(top_srcdir)/src/common_audio/signal_processing_library/main/interface \
-				       -I$(top_srcdir)/src/common_audio/vad/main/interface \
-				       -I$(top_srcdir)/src/system_wrappers/interface \
-				       -I$(top_srcdir)/src/modules/audio_processing/utility \
-				       -I$(top_srcdir)/src/modules/audio_processing/ns/interface \
-				       -I$(top_srcdir)/src/modules/audio_processing/aec/interface \
-				       -I$(top_srcdir)/src/modules/audio_processing/aecm/interface \
-				       -I$(top_srcdir)/src/modules/audio_processing/agc/interface
-libwebrtc_audio_processing_la_LIBADD = $(top_builddir)/src/system_wrappers/libsystem_wrappers.la \
-				       $(top_builddir)/src/common_audio/signal_processing_library/libspl.la \
-				       $(top_builddir)/src/common_audio/vad/libvad.la \
-				       $(top_builddir)/src/modules/audio_processing/utility/libapm_util.la \
-				       $(top_builddir)/src/modules/audio_processing/ns/$(NS_LIB).la \
-				       $(top_builddir)/src/modules/audio_processing/aec/libaec.la \
-				       $(top_builddir)/src/modules/audio_processing/aecm/libaecm.la \
-				       $(top_builddir)/src/modules/audio_processing/agc/libagc.la
-libwebrtc_audio_processing_la_LDFLAGS = $(AM_LDFLAGS) -version-info $(LIBWEBRTC_AUDIO_PROCESSING_VERSION_INFO)

src/modules/audio_processing/OWNERS

@@ -1,2 +0,0 @@
-andrew@webrtc.org
-bjornv@webrtc.org

src/modules/audio_processing/aec/Makefile.am

@@ -1,16 +0,0 @@
-noinst_LTLIBRARIES = libaec.la
-
-libaec_la_SOURCES = interface/echo_cancellation.h \
-		    echo_cancellation.c \
-		    aec_core.h \
-		    aec_core.c \
-		    aec_core_sse2.c \
-		    aec_rdft.h \
-		    aec_rdft.c \
-		    aec_rdft_sse2.c \
-		    resampler.h \
-		    resampler.c
-libaec_la_CFLAGS = $(AM_CFLAGS) $(COMMON_CFLAGS) \
-		   -I$(top_srcdir)/src/common_audio/signal_processing_library/main/interface \
-		   -I$(top_srcdir)/src/system_wrappers/interface \
-		   -I$(top_srcdir)/src/modules/audio_processing/utility

src/modules/audio_processing/aec/aec.gypi

@@ -1,40 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'targets': [
-    {
-      'target_name': 'aec',
-      'type': '<(library)',
-      'dependencies': [
-        '<(webrtc_root)/common_audio/common_audio.gyp:spl',
-        'apm_util'
-      ],
-      'include_dirs': [
-        'interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'interface',
-        ],
-      },
-      'sources': [
-        'interface/echo_cancellation.h',
-        'echo_cancellation.c',
-        'aec_core.h',
-        'aec_core.c',
-        'aec_core_sse2.c',
-        'aec_rdft.h',
-        'aec_rdft.c',
-        'aec_rdft_sse2.c',
-        'resampler.h',
-        'resampler.c',
-      ],
-    },
-  ],
-}

src/modules/audio_processing/aec/aec_core.h

@@ -1,181 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Specifies the interface for the AEC core.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
-
-#include <stdio.h>
-
-#include "signal_processing_library.h"
-#include "typedefs.h"
-
-//#define AEC_DEBUG // for recording files
-
-#define FRAME_LEN 80
-#define PART_LEN 64 // Length of partition
-#define PART_LEN1 (PART_LEN + 1) // Unique fft coefficients
-#define PART_LEN2 (PART_LEN * 2) // Length of partition * 2
-#define NR_PART 12 // Number of partitions
-#define FILT_LEN (PART_LEN * NR_PART) // Filter length
-#define FILT_LEN2 (FILT_LEN * 2) // Double filter length
-#define FAR_BUF_LEN (FILT_LEN2 * 2)
-#define PREF_BAND_SIZE 24
-
-#define BLOCKL_MAX FRAME_LEN
-// Maximum delay in fixed point delay estimator, used for logging
-enum {kMaxDelay = 100};
-
-typedef float complex_t[2];
-// For performance reasons, some arrays of complex numbers are replaced by twice
-// as long arrays of float, all the real parts followed by all the imaginary
-// ones (complex_t[SIZE] -> float[2][SIZE]). This allows SIMD optimizations and
-// is better than two arrays (one for the real parts and one for the imaginary
-// parts) as this other way would require two pointers instead of one and cause
-// extra register spilling. This also allows the offsets to be calculated at
-// compile time.
-
-// Metrics
-enum {offsetLevel = -100};
-
-typedef struct {
-    float sfrsum;
-    int sfrcounter;
-    float framelevel;
-    float frsum;
-    int frcounter;
-    float minlevel;
-    float averagelevel;
-} power_level_t;
-
-typedef struct {
-    float instant;
-    float average;
-    float min;
-    float max;
-    float sum;
-    float hisum;
-    float himean;
-    int counter;
-    int hicounter;
-} stats_t;
-
-typedef struct {
-    int farBufWritePos, farBufReadPos;
-
-    int knownDelay;
-    int inSamples, outSamples;
-    int delayEstCtr;
-
-    void *farFrBuf, *nearFrBuf, *outFrBuf;
-
-    void *nearFrBufH;
-    void *outFrBufH;
-
-    float xBuf[PART_LEN2]; // farend
-    float dBuf[PART_LEN2]; // nearend
-    float eBuf[PART_LEN2]; // error
-
-    float dBufH[PART_LEN2]; // nearend
-
-    float xPow[PART_LEN1];
-    float dPow[PART_LEN1];
-    float dMinPow[PART_LEN1];
-    float dInitMinPow[PART_LEN1];
-    float *noisePow;
-
-    float xfBuf[2][NR_PART * PART_LEN1]; // farend fft buffer
-    float wfBuf[2][NR_PART * PART_LEN1]; // filter fft
-    complex_t sde[PART_LEN1]; // cross-psd of nearend and error
-    complex_t sxd[PART_LEN1]; // cross-psd of farend and nearend
-    complex_t xfwBuf[NR_PART * PART_LEN1]; // farend windowed fft buffer
-
-    float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1]; // far, near and error psd
-    float hNs[PART_LEN1];
-    float hNlFbMin, hNlFbLocalMin;
-    float hNlXdAvgMin;
-    int hNlNewMin, hNlMinCtr;
-    float overDrive, overDriveSm;
-    float targetSupp, minOverDrive;
-    float outBuf[PART_LEN];
-    int delayIdx;
-
-    short stNearState, echoState;
-    short divergeState;
-
-    int xfBufBlockPos;
-
-    short farBuf[FILT_LEN2 * 2];
-
-    short mult; // sampling frequency multiple
-    int sampFreq;
-    WebRtc_UWord32 seed;
-
-    float mu; // stepsize
-    float errThresh; // error threshold
-
-    int noiseEstCtr;
-
-    power_level_t farlevel;
-    power_level_t nearlevel;
-    power_level_t linoutlevel;
-    power_level_t nlpoutlevel;
-
-    int metricsMode;
-    int stateCounter;
-    stats_t erl;
-    stats_t erle;
-    stats_t aNlp;
-    stats_t rerl;
-
-    // Quantities to control H band scaling for SWB input
-    int freq_avg_ic;         //initial bin for averaging nlp gain
-    int flag_Hband_cn;      //for comfort noise
-    float cn_scale_Hband;   //scale for comfort noise in H band
-
-    int delay_histogram[kMaxDelay];
-    int delay_logging_enabled;
-    void* delay_estimator;
-
-#ifdef AEC_DEBUG
-    FILE *farFile;
-    FILE *nearFile;
-    FILE *outFile;
-    FILE *outLpFile;
-#endif
-} aec_t;
-
-typedef void (*WebRtcAec_FilterFar_t)(aec_t *aec, float yf[2][PART_LEN1]);
-extern WebRtcAec_FilterFar_t WebRtcAec_FilterFar;
-typedef void (*WebRtcAec_ScaleErrorSignal_t)(aec_t *aec, float ef[2][PART_LEN1]);
-extern WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal;
-typedef void (*WebRtcAec_FilterAdaptation_t)
-  (aec_t *aec, float *fft, float ef[2][PART_LEN1]);
-extern WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation;
-typedef void (*WebRtcAec_OverdriveAndSuppress_t)
-  (aec_t *aec, float hNl[PART_LEN1], const float hNlFb, float efw[2][PART_LEN1]);
-extern WebRtcAec_OverdriveAndSuppress_t WebRtcAec_OverdriveAndSuppress;
-
-int WebRtcAec_CreateAec(aec_t **aec);
-int WebRtcAec_FreeAec(aec_t *aec);
-int WebRtcAec_InitAec(aec_t *aec, int sampFreq);
-void WebRtcAec_InitAec_SSE2(void);
-
-void WebRtcAec_InitMetrics(aec_t *aec);
-void WebRtcAec_ProcessFrame(aec_t *aec, const short *farend,
-                       const short *nearend, const short *nearendH,
-                       short *out, short *outH,
-                       int knownDelay);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
-

src/modules/audio_processing/aec/aec_core_sse2.c

@@ -1,417 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * The core AEC algorithm, SSE2 version of speed-critical functions.
- */
-
-#include "typedefs.h"
-
-#if defined(WEBRTC_USE_SSE2)
-#include <emmintrin.h>
-#include <math.h>
-
-#include "aec_core.h"
-#include "aec_rdft.h"
-
-__inline static float MulRe(float aRe, float aIm, float bRe, float bIm)
-{
-  return aRe * bRe - aIm * bIm;
-}
-
-__inline static float MulIm(float aRe, float aIm, float bRe, float bIm)
-{
-  return aRe * bIm + aIm * bRe;
-}
-
-static void FilterFarSSE2(aec_t *aec, float yf[2][PART_LEN1])
-{
-  int i;
-  for (i = 0; i < NR_PART; i++) {
-    int j;
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
-    int pos = i * PART_LEN1;
-    // Check for wrap
-    if (i + aec->xfBufBlockPos >= NR_PART) {
-      xPos -= NR_PART*(PART_LEN1);
-    }
-
-    // vectorized code (four at once)
-    for (j = 0; j + 3 < PART_LEN1; j += 4) {
-      const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
-      const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
-      const __m128 wfBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
-      const __m128 wfBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
-      const __m128 yf_re = _mm_loadu_ps(&yf[0][j]);
-      const __m128 yf_im = _mm_loadu_ps(&yf[1][j]);
-      const __m128 a = _mm_mul_ps(xfBuf_re, wfBuf_re);
-      const __m128 b = _mm_mul_ps(xfBuf_im, wfBuf_im);
-      const __m128 c = _mm_mul_ps(xfBuf_re, wfBuf_im);
-      const __m128 d = _mm_mul_ps(xfBuf_im, wfBuf_re);
-      const __m128 e = _mm_sub_ps(a, b);
-      const __m128 f = _mm_add_ps(c, d);
-      const __m128 g = _mm_add_ps(yf_re, e);
-      const __m128 h = _mm_add_ps(yf_im, f);
-      _mm_storeu_ps(&yf[0][j], g);
-      _mm_storeu_ps(&yf[1][j], h);
-    }
-    // scalar code for the remaining items.
-    for (; j < PART_LEN1; j++) {
-      yf[0][j] += MulRe(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
-      yf[1][j] += MulIm(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
-    }
-  }
-}
-
-static void ScaleErrorSignalSSE2(aec_t *aec, float ef[2][PART_LEN1])
-{
-  const __m128 k1e_10f = _mm_set1_ps(1e-10f);
-  const __m128 kThresh = _mm_set1_ps(aec->errThresh);
-  const __m128 kMu = _mm_set1_ps(aec->mu);
-
-  int i;
-  // vectorized code (four at once)
-  for (i = 0; i + 3 < PART_LEN1; i += 4) {
-    const __m128 xPow = _mm_loadu_ps(&aec->xPow[i]);
-    const __m128 ef_re_base = _mm_loadu_ps(&ef[0][i]);
-    const __m128 ef_im_base = _mm_loadu_ps(&ef[1][i]);
-
-    const __m128 xPowPlus = _mm_add_ps(xPow, k1e_10f);
-    __m128 ef_re = _mm_div_ps(ef_re_base, xPowPlus);
-    __m128 ef_im = _mm_div_ps(ef_im_base, xPowPlus);
-    const __m128 ef_re2 = _mm_mul_ps(ef_re, ef_re);
-    const __m128 ef_im2 = _mm_mul_ps(ef_im, ef_im);
-    const __m128 ef_sum2 = _mm_add_ps(ef_re2, ef_im2);
-    const __m128 absEf = _mm_sqrt_ps(ef_sum2);
-    const __m128 bigger = _mm_cmpgt_ps(absEf, kThresh);
-    __m128 absEfPlus = _mm_add_ps(absEf, k1e_10f);
-    const __m128 absEfInv = _mm_div_ps(kThresh, absEfPlus);
-    __m128 ef_re_if = _mm_mul_ps(ef_re, absEfInv);
-    __m128 ef_im_if = _mm_mul_ps(ef_im, absEfInv);
-    ef_re_if = _mm_and_ps(bigger, ef_re_if);
-    ef_im_if = _mm_and_ps(bigger, ef_im_if);
-    ef_re = _mm_andnot_ps(bigger, ef_re);
-    ef_im = _mm_andnot_ps(bigger, ef_im);
-    ef_re = _mm_or_ps(ef_re, ef_re_if);
-    ef_im = _mm_or_ps(ef_im, ef_im_if);
-    ef_re = _mm_mul_ps(ef_re, kMu);
-    ef_im = _mm_mul_ps(ef_im, kMu);
-
-    _mm_storeu_ps(&ef[0][i], ef_re);
-    _mm_storeu_ps(&ef[1][i], ef_im);
-  }
-  // scalar code for the remaining items.
-  for (; i < (PART_LEN1); i++) {
-    float absEf;
-    ef[0][i] /= (aec->xPow[i] + 1e-10f);
-    ef[1][i] /= (aec->xPow[i] + 1e-10f);
-    absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
-
-    if (absEf > aec->errThresh) {
-      absEf = aec->errThresh / (absEf + 1e-10f);
-      ef[0][i] *= absEf;
-      ef[1][i] *= absEf;
-    }
-
-    // Stepsize factor
-    ef[0][i] *= aec->mu;
-    ef[1][i] *= aec->mu;
-  }
-}
-
-static void FilterAdaptationSSE2(aec_t *aec, float *fft, float ef[2][PART_LEN1]) {
-  int i, j;
-  for (i = 0; i < NR_PART; i++) {
-    int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
-    int pos = i * PART_LEN1;
-    // Check for wrap
-    if (i + aec->xfBufBlockPos >= NR_PART) {
-      xPos -= NR_PART * PART_LEN1;
-    }
-
-    // Process the whole array...
-    for (j = 0; j < PART_LEN; j+= 4) {
-      // Load xfBuf and ef.
-      const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
-      const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
-      const __m128 ef_re = _mm_loadu_ps(&ef[0][j]);
-      const __m128 ef_im = _mm_loadu_ps(&ef[1][j]);
-      // Calculate the product of conjugate(xfBuf) by ef.
-      //   re(conjugate(a) * b) = aRe * bRe + aIm * bIm
-      //   im(conjugate(a) * b)=  aRe * bIm - aIm * bRe
-      const __m128 a = _mm_mul_ps(xfBuf_re, ef_re);
-      const __m128 b = _mm_mul_ps(xfBuf_im, ef_im);
-      const __m128 c = _mm_mul_ps(xfBuf_re, ef_im);
-      const __m128 d = _mm_mul_ps(xfBuf_im, ef_re);
-      const __m128 e = _mm_add_ps(a, b);
-      const __m128 f = _mm_sub_ps(c, d);
-      // Interleave real and imaginary parts.
-      const __m128 g = _mm_unpacklo_ps(e, f);
-      const __m128 h = _mm_unpackhi_ps(e, f);
-      // Store
-      _mm_storeu_ps(&fft[2*j + 0], g);
-      _mm_storeu_ps(&fft[2*j + 4], h);
-    }
-    // ... and fixup the first imaginary entry.
-    fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
-                   -aec->xfBuf[1][xPos + PART_LEN],
-                   ef[0][PART_LEN], ef[1][PART_LEN]);
-
-    aec_rdft_inverse_128(fft);
-    memset(fft + PART_LEN, 0, sizeof(float)*PART_LEN);
-
-    // fft scaling
-    {
-      float scale = 2.0f / PART_LEN2;
-      const __m128 scale_ps = _mm_load_ps1(&scale);
-      for (j = 0; j < PART_LEN; j+=4) {
-        const __m128 fft_ps = _mm_loadu_ps(&fft[j]);
-        const __m128 fft_scale = _mm_mul_ps(fft_ps, scale_ps);
-        _mm_storeu_ps(&fft[j], fft_scale);
-      }
-    }
-    aec_rdft_forward_128(fft);
-
-    {
-      float wt1 = aec->wfBuf[1][pos];
-      aec->wfBuf[0][pos + PART_LEN] += fft[1];
-      for (j = 0; j < PART_LEN; j+= 4) {
-        __m128 wtBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
-        __m128 wtBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
-        const __m128 fft0 = _mm_loadu_ps(&fft[2 * j + 0]);
-        const __m128 fft4 = _mm_loadu_ps(&fft[2 * j + 4]);
-        const __m128 fft_re = _mm_shuffle_ps(fft0, fft4, _MM_SHUFFLE(2, 0, 2 ,0));
-        const __m128 fft_im = _mm_shuffle_ps(fft0, fft4, _MM_SHUFFLE(3, 1, 3 ,1));
-        wtBuf_re = _mm_add_ps(wtBuf_re, fft_re);
-        wtBuf_im = _mm_add_ps(wtBuf_im, fft_im);
-        _mm_storeu_ps(&aec->wfBuf[0][pos + j], wtBuf_re);
-        _mm_storeu_ps(&aec->wfBuf[1][pos + j], wtBuf_im);
-      }
-      aec->wfBuf[1][pos] = wt1;
-    }
-  }
-}
-
-static __m128 mm_pow_ps(__m128 a, __m128 b)
-{
-  // a^b = exp2(b * log2(a))
-  //   exp2(x) and log2(x) are calculated using polynomial approximations.
-  __m128 log2_a, b_log2_a, a_exp_b;
-
-  // Calculate log2(x), x = a.
-  {
-    // To calculate log2(x), we decompose x like this:
-    //   x = y * 2^n
-    //     n is an integer
-    //     y is in the [1.0, 2.0) range
-    //
-    //   log2(x) = log2(y) + n
-    //     n       can be evaluated by playing with float representation.
-    //     log2(y) in a small range can be approximated, this code uses an order
-    //             five polynomial approximation. The coefficients have been
-    //             estimated with the Remez algorithm and the resulting
-    //             polynomial has a maximum relative error of 0.00086%.
-
-    // Compute n.
-    //    This is done by masking the exponent, shifting it into the top bit of
-    //    the mantissa, putting eight into the biased exponent (to shift/
-    //    compensate the fact that the exponent has been shifted in the top/
-    //    fractional part and finally getting rid of the implicit leading one
-    //    from the mantissa by substracting it out.
-    static const ALIGN16_BEG int float_exponent_mask[4] ALIGN16_END =
-        {0x7F800000, 0x7F800000, 0x7F800000, 0x7F800000};
-    static const ALIGN16_BEG int eight_biased_exponent[4] ALIGN16_END =
-        {0x43800000, 0x43800000, 0x43800000, 0x43800000};
-    static const ALIGN16_BEG int implicit_leading_one[4] ALIGN16_END =
-        {0x43BF8000, 0x43BF8000, 0x43BF8000, 0x43BF8000};
-    static const int shift_exponent_into_top_mantissa = 8;
-    const __m128 two_n = _mm_and_ps(a, *((__m128 *)float_exponent_mask));
-    const __m128 n_1 = _mm_castsi128_ps(_mm_srli_epi32(_mm_castps_si128(two_n),
-        shift_exponent_into_top_mantissa));
-    const __m128 n_0 = _mm_or_ps(n_1, *((__m128 *)eight_biased_exponent));
-    const __m128 n   = _mm_sub_ps(n_0,  *((__m128 *)implicit_leading_one));
-
-    // Compute y.
-    static const ALIGN16_BEG int mantissa_mask[4] ALIGN16_END =
-        {0x007FFFFF, 0x007FFFFF, 0x007FFFFF, 0x007FFFFF};
-    static const ALIGN16_BEG int zero_biased_exponent_is_one[4] ALIGN16_END =
-        {0x3F800000, 0x3F800000, 0x3F800000, 0x3F800000};
-    const __m128 mantissa = _mm_and_ps(a, *((__m128 *)mantissa_mask));
-    const __m128 y        = _mm_or_ps(
-        mantissa,  *((__m128 *)zero_biased_exponent_is_one));
-
-    // Approximate log2(y) ~= (y - 1) * pol5(y).
-    //    pol5(y) = C5 * y^5 + C4 * y^4 + C3 * y^3 + C2 * y^2 + C1 * y + C0
-    static const ALIGN16_BEG float ALIGN16_END C5[4] =
-        {-3.4436006e-2f, -3.4436006e-2f, -3.4436006e-2f, -3.4436006e-2f};
-    static const ALIGN16_BEG float ALIGN16_END C4[4] =
-        {3.1821337e-1f, 3.1821337e-1f, 3.1821337e-1f, 3.1821337e-1f};
-    static const ALIGN16_BEG float ALIGN16_END C3[4] =
-        {-1.2315303f, -1.2315303f, -1.2315303f, -1.2315303f};
-    static const ALIGN16_BEG float ALIGN16_END C2[4] =
-        {2.5988452f, 2.5988452f, 2.5988452f, 2.5988452f};
-    static const ALIGN16_BEG float ALIGN16_END C1[4] =
-        {-3.3241990f, -3.3241990f, -3.3241990f, -3.3241990f};
-    static const ALIGN16_BEG float ALIGN16_END C0[4] =
-        {3.1157899f, 3.1157899f, 3.1157899f, 3.1157899f};
-    const __m128 pol5_y_0 = _mm_mul_ps(y,        *((__m128 *)C5));
-    const __m128 pol5_y_1 = _mm_add_ps(pol5_y_0, *((__m128 *)C4));
-    const __m128 pol5_y_2 = _mm_mul_ps(pol5_y_1, y);
-    const __m128 pol5_y_3 = _mm_add_ps(pol5_y_2, *((__m128 *)C3));
-    const __m128 pol5_y_4 = _mm_mul_ps(pol5_y_3, y);
-    const __m128 pol5_y_5 = _mm_add_ps(pol5_y_4, *((__m128 *)C2));
-    const __m128 pol5_y_6 = _mm_mul_ps(pol5_y_5, y);
-    const __m128 pol5_y_7 = _mm_add_ps(pol5_y_6, *((__m128 *)C1));
-    const __m128 pol5_y_8 = _mm_mul_ps(pol5_y_7, y);
-    const __m128 pol5_y   = _mm_add_ps(pol5_y_8, *((__m128 *)C0));
-    const __m128 y_minus_one = _mm_sub_ps(
-        y, *((__m128 *)zero_biased_exponent_is_one));
-    const __m128 log2_y = _mm_mul_ps(y_minus_one ,  pol5_y);
-
-    // Combine parts.
-    log2_a = _mm_add_ps(n, log2_y);
-  }
-
-  // b * log2(a)
-  b_log2_a = _mm_mul_ps(b, log2_a);
-
-  // Calculate exp2(x), x = b * log2(a).
-  {
-    // To calculate 2^x, we decompose x like this:
-    //   x = n + y
-    //     n is an integer, the value of x - 0.5 rounded down, therefore
-    //     y is in the [0.5, 1.5) range
-    //
-    //   2^x = 2^n * 2^y
-    //     2^n can be evaluated by playing with float representation.
-    //     2^y in a small range can be approximated, this code uses an order two
-    //         polynomial approximation. The coefficients have been estimated
-    //         with the Remez algorithm and the resulting polynomial has a
-    //         maximum relative error of 0.17%.
-
-    // To avoid over/underflow, we reduce the range of input to ]-127, 129].
-    static const ALIGN16_BEG float max_input[4] ALIGN16_END =
-        {129.f, 129.f, 129.f, 129.f};
-    static const ALIGN16_BEG float min_input[4] ALIGN16_END =
-        {-126.99999f, -126.99999f, -126.99999f, -126.99999f};
-    const __m128 x_min = _mm_min_ps(b_log2_a, *((__m128 *)max_input));
-    const __m128 x_max = _mm_max_ps(x_min,    *((__m128 *)min_input));
-    // Compute n.
-    static const ALIGN16_BEG float half[4] ALIGN16_END =
-        {0.5f, 0.5f, 0.5f, 0.5f};
-    const __m128  x_minus_half = _mm_sub_ps(x_max, *((__m128 *)half));
-    const __m128i x_minus_half_floor = _mm_cvtps_epi32(x_minus_half);
-    // Compute 2^n.
-    static const ALIGN16_BEG int float_exponent_bias[4] ALIGN16_END =
-        {127, 127, 127, 127};
-    static const int float_exponent_shift = 23;
-    const __m128i two_n_exponent = _mm_add_epi32(
-        x_minus_half_floor, *((__m128i *)float_exponent_bias));
-    const __m128  two_n = _mm_castsi128_ps(_mm_slli_epi32(
-        two_n_exponent, float_exponent_shift));
-    // Compute y.
-    const __m128 y = _mm_sub_ps(x_max, _mm_cvtepi32_ps(x_minus_half_floor));
-    // Approximate 2^y ~= C2 * y^2 + C1 * y + C0.
-    static const ALIGN16_BEG float C2[4] ALIGN16_END =
-        {3.3718944e-1f, 3.3718944e-1f, 3.3718944e-1f, 3.3718944e-1f};
-    static const ALIGN16_BEG float C1[4] ALIGN16_END =
-        {6.5763628e-1f, 6.5763628e-1f, 6.5763628e-1f, 6.5763628e-1f};
-    static const ALIGN16_BEG float C0[4] ALIGN16_END =
-        {1.0017247f, 1.0017247f, 1.0017247f, 1.0017247f};
-    const __m128 exp2_y_0 = _mm_mul_ps(y,        *((__m128 *)C2));
-    const __m128 exp2_y_1 = _mm_add_ps(exp2_y_0, *((__m128 *)C1));
-    const __m128 exp2_y_2 = _mm_mul_ps(exp2_y_1, y);
-    const __m128 exp2_y   = _mm_add_ps(exp2_y_2, *((__m128 *)C0));
-
-    // Combine parts.
-    a_exp_b = _mm_mul_ps(exp2_y, two_n);
-  }
-  return a_exp_b;
-}
-
-extern const float WebRtcAec_weightCurve[65];
-extern const float WebRtcAec_overDriveCurve[65];
-
-static void OverdriveAndSuppressSSE2(aec_t *aec, float hNl[PART_LEN1],
-                                     const float hNlFb,
-                                     float efw[2][PART_LEN1]) {
-  int i;
-  const __m128 vec_hNlFb = _mm_set1_ps(hNlFb);
-  const __m128 vec_one = _mm_set1_ps(1.0f);
-  const __m128 vec_minus_one = _mm_set1_ps(-1.0f);
-  const __m128 vec_overDriveSm = _mm_set1_ps(aec->overDriveSm);
-  // vectorized code (four at once)
-  for (i = 0; i + 3 < PART_LEN1; i+=4) {
-    // Weight subbands
-    __m128 vec_hNl = _mm_loadu_ps(&hNl[i]);
-    const __m128 vec_weightCurve = _mm_loadu_ps(&WebRtcAec_weightCurve[i]);
-    const __m128 bigger = _mm_cmpgt_ps(vec_hNl, vec_hNlFb);
-    const __m128 vec_weightCurve_hNlFb = _mm_mul_ps(
-        vec_weightCurve, vec_hNlFb);
-    const __m128 vec_one_weightCurve = _mm_sub_ps(vec_one, vec_weightCurve);
-    const __m128 vec_one_weightCurve_hNl = _mm_mul_ps(
-        vec_one_weightCurve, vec_hNl);
-    const __m128 vec_if0 = _mm_andnot_ps(bigger, vec_hNl);
-    const __m128 vec_if1 = _mm_and_ps(
-        bigger, _mm_add_ps(vec_weightCurve_hNlFb, vec_one_weightCurve_hNl));
-    vec_hNl = _mm_or_ps(vec_if0, vec_if1);
-
-    {
-      const __m128 vec_overDriveCurve = _mm_loadu_ps(
-          &WebRtcAec_overDriveCurve[i]);
-      const __m128 vec_overDriveSm_overDriveCurve = _mm_mul_ps(
-          vec_overDriveSm, vec_overDriveCurve);
-      vec_hNl = mm_pow_ps(vec_hNl, vec_overDriveSm_overDriveCurve);
-      _mm_storeu_ps(&hNl[i], vec_hNl);
-    }
-
-    // Suppress error signal
-    {
-      __m128 vec_efw_re = _mm_loadu_ps(&efw[0][i]);
-      __m128 vec_efw_im = _mm_loadu_ps(&efw[1][i]);
-      vec_efw_re = _mm_mul_ps(vec_efw_re, vec_hNl);
-      vec_efw_im = _mm_mul_ps(vec_efw_im, vec_hNl);
-
-      // Ooura fft returns incorrect sign on imaginary component. It matters
-      // here because we are making an additive change with comfort noise.
-      vec_efw_im = _mm_mul_ps(vec_efw_im, vec_minus_one);
-      _mm_storeu_ps(&efw[0][i], vec_efw_re);
-      _mm_storeu_ps(&efw[1][i], vec_efw_im);
-    }
-  }
-  // scalar code for the remaining items.
-  for (; i < PART_LEN1; i++) {
-    // Weight subbands
-    if (hNl[i] > hNlFb) {
-      hNl[i] = WebRtcAec_weightCurve[i] * hNlFb +
-          (1 - WebRtcAec_weightCurve[i]) * hNl[i];
-    }
-    hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
-
-    // Suppress error signal
-    efw[0][i] *= hNl[i];
-    efw[1][i] *= hNl[i];
-
-    // Ooura fft returns incorrect sign on imaginary component. It matters
-    // here because we are making an additive change with comfort noise.
-    efw[1][i] *= -1;
-  }
-}
-
-void WebRtcAec_InitAec_SSE2(void) {
-  WebRtcAec_FilterFar = FilterFarSSE2;
-  WebRtcAec_ScaleErrorSignal = ScaleErrorSignalSSE2;
-  WebRtcAec_FilterAdaptation = FilterAdaptationSSE2;
-  WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
-}
-
-#endif   // WEBRTC_USE_SSE2

src/modules/audio_processing/aec/aec_rdft.h

@@ -1,57 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
-
-// These intrinsics were unavailable before VS 2008.
-// TODO(andrew): move to a common file.
-#if defined(_MSC_VER) && _MSC_VER < 1500
-#include <emmintrin.h>
-static __inline __m128 _mm_castsi128_ps(__m128i a) { return *(__m128*)&a; }
-static __inline __m128i _mm_castps_si128(__m128 a) { return *(__m128i*)&a; }
-#endif
-
-#ifdef _MSC_VER /* visual c++ */
-# define ALIGN16_BEG __declspec(align(16))
-# define ALIGN16_END
-#else /* gcc or icc */
-# define ALIGN16_BEG
-# define ALIGN16_END __attribute__((aligned(16)))
-#endif
-
-// constants shared by all paths (C, SSE2).
-extern float rdft_w[64];
-// constants used by the C path.
-extern float rdft_wk3ri_first[32];
-extern float rdft_wk3ri_second[32];
-// constants used by SSE2 but initialized in C path.
-extern float rdft_wk1r[32];
-extern float rdft_wk2r[32];
-extern float rdft_wk3r[32];
-extern float rdft_wk1i[32];
-extern float rdft_wk2i[32];
-extern float rdft_wk3i[32];
-extern float cftmdl_wk1r[4];
-
-// code path selection function pointers
-typedef void (*rft_sub_128_t)(float *a);
-extern rft_sub_128_t rftfsub_128;
-extern rft_sub_128_t rftbsub_128;
-extern rft_sub_128_t cft1st_128;
-extern rft_sub_128_t cftmdl_128;
-
-// entry points
-void aec_rdft_init(void);
-void aec_rdft_init_sse2(void);
-void aec_rdft_forward_128(float *a);
-void aec_rdft_inverse_128(float *a);
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_

src/modules/audio_processing/aec/aec_rdft_sse2.c

@@ -1,431 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "typedefs.h"
-
-#if defined(WEBRTC_USE_SSE2)
-#include <emmintrin.h>
-
-#include "aec_rdft.h"
-
-static const ALIGN16_BEG float ALIGN16_END k_swap_sign[4] =
-  {-1.f, 1.f, -1.f, 1.f};
-
-static void cft1st_128_SSE2(float *a) {
-  const __m128 mm_swap_sign = _mm_load_ps(k_swap_sign);
-  int j, k2;
-
-  for (k2 = 0, j = 0; j < 128; j += 16, k2 += 4) {
-          __m128 a00v   = _mm_loadu_ps(&a[j +  0]);
-          __m128 a04v   = _mm_loadu_ps(&a[j +  4]);
-          __m128 a08v   = _mm_loadu_ps(&a[j +  8]);
-          __m128 a12v   = _mm_loadu_ps(&a[j + 12]);
-          __m128 a01v   = _mm_shuffle_ps(a00v, a08v, _MM_SHUFFLE(1, 0, 1 ,0));
-          __m128 a23v   = _mm_shuffle_ps(a00v, a08v, _MM_SHUFFLE(3, 2, 3 ,2));
-          __m128 a45v   = _mm_shuffle_ps(a04v, a12v, _MM_SHUFFLE(1, 0, 1 ,0));
-          __m128 a67v   = _mm_shuffle_ps(a04v, a12v, _MM_SHUFFLE(3, 2, 3 ,2));
-
-    const __m128 wk1rv  = _mm_load_ps(&rdft_wk1r[k2]);
-    const __m128 wk1iv  = _mm_load_ps(&rdft_wk1i[k2]);
-    const __m128 wk2rv  = _mm_load_ps(&rdft_wk2r[k2]);
-    const __m128 wk2iv  = _mm_load_ps(&rdft_wk2i[k2]);
-    const __m128 wk3rv  = _mm_load_ps(&rdft_wk3r[k2]);
-    const __m128 wk3iv  = _mm_load_ps(&rdft_wk3i[k2]);
-          __m128 x0v    = _mm_add_ps(a01v, a23v);
-    const __m128 x1v    = _mm_sub_ps(a01v, a23v);
-    const __m128 x2v    = _mm_add_ps(a45v, a67v);
-    const __m128 x3v    = _mm_sub_ps(a45v, a67v);
-          __m128 x0w;
-                 a01v   = _mm_add_ps(x0v, x2v);
-                 x0v    = _mm_sub_ps(x0v, x2v);
-                 x0w    = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
-    {
-      const __m128 a45_0v = _mm_mul_ps(wk2rv, x0v);
-      const __m128 a45_1v = _mm_mul_ps(wk2iv, x0w);
-                   a45v   = _mm_add_ps(a45_0v, a45_1v);
-    }
-    {
-            __m128 a23_0v, a23_1v;
-      const __m128 x3w    = _mm_shuffle_ps(x3v, x3v, _MM_SHUFFLE(2, 3, 0 ,1));
-      const __m128 x3s    = _mm_mul_ps(mm_swap_sign, x3w);
-                   x0v    = _mm_add_ps(x1v, x3s);
-                   x0w    = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
-                   a23_0v = _mm_mul_ps(wk1rv, x0v);
-                   a23_1v = _mm_mul_ps(wk1iv, x0w);
-                   a23v   = _mm_add_ps(a23_0v, a23_1v);
-
-                   x0v    = _mm_sub_ps(x1v, x3s);
-                   x0w    = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
-    }
-    {
-      const __m128 a67_0v = _mm_mul_ps(wk3rv, x0v);
-      const __m128 a67_1v = _mm_mul_ps(wk3iv, x0w);
-                   a67v   = _mm_add_ps(a67_0v, a67_1v);
-    }
-
-                 a00v   = _mm_shuffle_ps(a01v, a23v, _MM_SHUFFLE(1, 0, 1 ,0));
-                 a04v   = _mm_shuffle_ps(a45v, a67v, _MM_SHUFFLE(1, 0, 1 ,0));
-                 a08v   = _mm_shuffle_ps(a01v, a23v, _MM_SHUFFLE(3, 2, 3 ,2));
-                 a12v   = _mm_shuffle_ps(a45v, a67v, _MM_SHUFFLE(3, 2, 3 ,2));
-    _mm_storeu_ps(&a[j +  0], a00v);
-    _mm_storeu_ps(&a[j +  4], a04v);
-    _mm_storeu_ps(&a[j +  8], a08v);
-    _mm_storeu_ps(&a[j + 12], a12v);
-  }
-}
-
-static void cftmdl_128_SSE2(float *a) {
-  const int l = 8;
-  const __m128 mm_swap_sign = _mm_load_ps(k_swap_sign);
-  int j0;
-
-  __m128 wk1rv = _mm_load_ps(cftmdl_wk1r);
-  for (j0 = 0; j0 < l; j0 += 2) {
-    const __m128i a_00 = _mm_loadl_epi64((__m128i*)&a[j0 +  0]);
-    const __m128i a_08 = _mm_loadl_epi64((__m128i*)&a[j0 +  8]);
-    const __m128i a_32 = _mm_loadl_epi64((__m128i*)&a[j0 + 32]);
-    const __m128i a_40 = _mm_loadl_epi64((__m128i*)&a[j0 + 40]);
-    const __m128  a_00_32 = _mm_shuffle_ps(_mm_castsi128_ps(a_00),
-                                           _mm_castsi128_ps(a_32),
-                                           _MM_SHUFFLE(1, 0, 1 ,0));
-    const __m128  a_08_40 = _mm_shuffle_ps(_mm_castsi128_ps(a_08),
-                                           _mm_castsi128_ps(a_40),
-                                           _MM_SHUFFLE(1, 0, 1 ,0));
-          __m128  x0r0_0i0_0r1_x0i1 = _mm_add_ps(a_00_32, a_08_40);
-    const __m128  x1r0_1i0_1r1_x1i1 = _mm_sub_ps(a_00_32, a_08_40);
-
-    const __m128i a_16 = _mm_loadl_epi64((__m128i*)&a[j0 + 16]);
-    const __m128i a_24 = _mm_loadl_epi64((__m128i*)&a[j0 + 24]);
-    const __m128i a_48 = _mm_loadl_epi64((__m128i*)&a[j0 + 48]);
-    const __m128i a_56 = _mm_loadl_epi64((__m128i*)&a[j0 + 56]);
-    const __m128  a_16_48 = _mm_shuffle_ps(_mm_castsi128_ps(a_16),
-                                           _mm_castsi128_ps(a_48),
-                                           _MM_SHUFFLE(1, 0, 1 ,0));
-    const __m128  a_24_56 = _mm_shuffle_ps(_mm_castsi128_ps(a_24),
-                                           _mm_castsi128_ps(a_56),
-                                           _MM_SHUFFLE(1, 0, 1 ,0));
-    const __m128  x2r0_2i0_2r1_x2i1 = _mm_add_ps(a_16_48, a_24_56);
-    const __m128  x3r0_3i0_3r1_x3i1 = _mm_sub_ps(a_16_48, a_24_56);
-
-    const __m128  xx0 = _mm_add_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
-    const __m128  xx1 = _mm_sub_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
-
-    const __m128  x3i0_3r0_3i1_x3r1 = _mm_castsi128_ps(
-        _mm_shuffle_epi32(_mm_castps_si128(x3r0_3i0_3r1_x3i1),
-                          _MM_SHUFFLE(2, 3, 0, 1)));
-    const __m128  x3_swapped = _mm_mul_ps(mm_swap_sign, x3i0_3r0_3i1_x3r1);
-    const __m128  x1_x3_add = _mm_add_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
-    const __m128  x1_x3_sub = _mm_sub_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
-
-    const __m128 yy0 = _mm_shuffle_ps(x1_x3_add, x1_x3_sub,
-                                      _MM_SHUFFLE(2, 2, 2 ,2));
-    const __m128 yy1 = _mm_shuffle_ps(x1_x3_add, x1_x3_sub,
-                                      _MM_SHUFFLE(3, 3, 3 ,3));
-    const __m128 yy2 = _mm_mul_ps(mm_swap_sign, yy1);
-    const __m128 yy3 = _mm_add_ps(yy0, yy2);
-    const __m128 yy4 = _mm_mul_ps(wk1rv, yy3);
-
-    _mm_storel_epi64((__m128i*)&a[j0 +  0], _mm_castps_si128(xx0));
-    _mm_storel_epi64((__m128i*)&a[j0 + 32],
-                     _mm_shuffle_epi32(_mm_castps_si128(xx0),
-                                       _MM_SHUFFLE(3, 2, 3, 2)));
-
-    _mm_storel_epi64((__m128i*)&a[j0 + 16], _mm_castps_si128(xx1));
-    _mm_storel_epi64((__m128i*)&a[j0 + 48],
-                     _mm_shuffle_epi32(_mm_castps_si128(xx1),
-                                       _MM_SHUFFLE(2, 3, 2, 3)));
-    a[j0 + 48] = -a[j0 + 48];
-
-    _mm_storel_epi64((__m128i*)&a[j0 +  8], _mm_castps_si128(x1_x3_add));
-    _mm_storel_epi64((__m128i*)&a[j0 + 24], _mm_castps_si128(x1_x3_sub));
-
-    _mm_storel_epi64((__m128i*)&a[j0 + 40], _mm_castps_si128(yy4));
-    _mm_storel_epi64((__m128i*)&a[j0 + 56],
-                     _mm_shuffle_epi32(_mm_castps_si128(yy4),
-                     _MM_SHUFFLE(2, 3, 2, 3)));
-  }
-
-  {
-    int k = 64;
-    int k1 = 2;
-    int k2 = 2 * k1;
-    const __m128 wk2rv = _mm_load_ps(&rdft_wk2r[k2+0]);
-    const __m128 wk2iv = _mm_load_ps(&rdft_wk2i[k2+0]);
-    const __m128 wk1iv = _mm_load_ps(&rdft_wk1i[k2+0]);
-    const __m128 wk3rv = _mm_load_ps(&rdft_wk3r[k2+0]);
-    const __m128 wk3iv = _mm_load_ps(&rdft_wk3i[k2+0]);
-                 wk1rv = _mm_load_ps(&rdft_wk1r[k2+0]);
-    for (j0 = k; j0 < l + k; j0 += 2) {
-      const __m128i a_00 = _mm_loadl_epi64((__m128i*)&a[j0 +  0]);
-      const __m128i a_08 = _mm_loadl_epi64((__m128i*)&a[j0 +  8]);
-      const __m128i a_32 = _mm_loadl_epi64((__m128i*)&a[j0 + 32]);
-      const __m128i a_40 = _mm_loadl_epi64((__m128i*)&a[j0 + 40]);
-      const __m128 a_00_32 = _mm_shuffle_ps(_mm_castsi128_ps(a_00),
-                                            _mm_castsi128_ps(a_32),
-                                            _MM_SHUFFLE(1, 0, 1 ,0));
-      const __m128 a_08_40 = _mm_shuffle_ps(_mm_castsi128_ps(a_08),
-                                            _mm_castsi128_ps(a_40),
-                                            _MM_SHUFFLE(1, 0, 1 ,0));
-            __m128 x0r0_0i0_0r1_x0i1 = _mm_add_ps(a_00_32, a_08_40);
-      const __m128 x1r0_1i0_1r1_x1i1 = _mm_sub_ps(a_00_32, a_08_40);
-
-      const __m128i a_16 = _mm_loadl_epi64((__m128i*)&a[j0 + 16]);
-      const __m128i a_24 = _mm_loadl_epi64((__m128i*)&a[j0 + 24]);
-      const __m128i a_48 = _mm_loadl_epi64((__m128i*)&a[j0 + 48]);
-      const __m128i a_56 = _mm_loadl_epi64((__m128i*)&a[j0 + 56]);
-      const __m128 a_16_48 = _mm_shuffle_ps(_mm_castsi128_ps(a_16),
-                                            _mm_castsi128_ps(a_48),
-                                            _MM_SHUFFLE(1, 0, 1 ,0));
-      const __m128 a_24_56 = _mm_shuffle_ps(_mm_castsi128_ps(a_24),
-                                            _mm_castsi128_ps(a_56),
-                                            _MM_SHUFFLE(1, 0, 1 ,0));
-      const __m128 x2r0_2i0_2r1_x2i1 = _mm_add_ps(a_16_48, a_24_56);
-      const __m128 x3r0_3i0_3r1_x3i1 = _mm_sub_ps(a_16_48, a_24_56);
-
-      const __m128 xx = _mm_add_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
-      const __m128 xx1 = _mm_sub_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
-      const __m128 xx2 = _mm_mul_ps(xx1 , wk2rv);
-      const __m128 xx3 = _mm_mul_ps(wk2iv,
-          _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(xx1),
-                                             _MM_SHUFFLE(2, 3, 0, 1))));
-      const __m128 xx4 = _mm_add_ps(xx2, xx3);
-
-      const __m128  x3i0_3r0_3i1_x3r1 =  _mm_castsi128_ps(
-          _mm_shuffle_epi32(_mm_castps_si128(x3r0_3i0_3r1_x3i1),
-                            _MM_SHUFFLE(2, 3, 0, 1)));
-      const __m128  x3_swapped = _mm_mul_ps(mm_swap_sign, x3i0_3r0_3i1_x3r1);
-      const __m128  x1_x3_add = _mm_add_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
-      const __m128  x1_x3_sub = _mm_sub_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
-
-      const __m128 xx10 = _mm_mul_ps(x1_x3_add, wk1rv);
-      const __m128 xx11 = _mm_mul_ps(wk1iv,
-          _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(x1_x3_add),
-                                             _MM_SHUFFLE(2, 3, 0, 1))));
-      const __m128 xx12 = _mm_add_ps(xx10, xx11);
-
-      const __m128 xx20 = _mm_mul_ps(x1_x3_sub, wk3rv);
-      const __m128 xx21 = _mm_mul_ps(wk3iv,
-          _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(x1_x3_sub),
-                           _MM_SHUFFLE(2, 3, 0, 1))));
-      const __m128 xx22 = _mm_add_ps(xx20, xx21);
-
-      _mm_storel_epi64((__m128i*)&a[j0 +  0], _mm_castps_si128(xx));
-      _mm_storel_epi64((__m128i*)&a[j0 + 32],
-                         _mm_shuffle_epi32(_mm_castps_si128(xx),
-                                           _MM_SHUFFLE(3, 2, 3, 2)));
-
-      _mm_storel_epi64((__m128i*)&a[j0 + 16], _mm_castps_si128(xx4));
-      _mm_storel_epi64((__m128i*)&a[j0 + 48],
-                        _mm_shuffle_epi32(_mm_castps_si128(xx4),
-                                          _MM_SHUFFLE(3, 2, 3, 2)));
-
-      _mm_storel_epi64((__m128i*)&a[j0 +  8], _mm_castps_si128(xx12));
-      _mm_storel_epi64((__m128i*)&a[j0 + 40],
-                       _mm_shuffle_epi32(_mm_castps_si128(xx12),
-                                         _MM_SHUFFLE(3, 2, 3, 2)));
-
-      _mm_storel_epi64((__m128i*)&a[j0 + 24], _mm_castps_si128(xx22));
-      _mm_storel_epi64((__m128i*)&a[j0 + 56],
-                       _mm_shuffle_epi32(_mm_castps_si128(xx22),
-                                         _MM_SHUFFLE(3, 2, 3, 2)));
-    }
-  }
-}
-
-static void rftfsub_128_SSE2(float *a) {
-  const float *c = rdft_w + 32;
-  int j1, j2, k1, k2;
-  float wkr, wki, xr, xi, yr, yi;
-
-  static const ALIGN16_BEG float ALIGN16_END k_half[4] =
-    {0.5f, 0.5f, 0.5f, 0.5f};
-  const __m128 mm_half = _mm_load_ps(k_half);
-
-  // Vectorized code (four at once).
-  //    Note: commented number are indexes for the first iteration of the loop.
-  for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
-    // Load 'wk'.
-    const __m128 c_j1 = _mm_loadu_ps(&c[     j1]);         //  1,  2,  3,  4,
-    const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]);         // 28, 29, 30, 31,
-    const __m128 wkrt = _mm_sub_ps(mm_half, c_k1);         // 28, 29, 30, 31,
-    const __m128 wkr_ =
-      _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
-    const __m128 wki_ = c_j1;                              //  1,  2,  3,  4,
-    // Load and shuffle 'a'.
-    const __m128 a_j2_0 = _mm_loadu_ps(&a[0   + j2]);  //   2,   3,   4,   5,
-    const __m128 a_j2_4 = _mm_loadu_ps(&a[4   + j2]);  //   6,   7,   8,   9,
-    const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]);  // 120, 121, 122, 123,
-    const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]);  // 124, 125, 126, 127,
-    const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
-                            _MM_SHUFFLE(2, 0, 2 ,0));  //   2,   4,   6,   8,
-    const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
-                            _MM_SHUFFLE(3, 1, 3 ,1));  //   3,   5,   7,   9,
-    const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
-                            _MM_SHUFFLE(0, 2, 0 ,2));  // 126, 124, 122, 120,
-    const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
-                            _MM_SHUFFLE(1, 3, 1 ,3));  // 127, 125, 123, 121,
-    // Calculate 'x'.
-    const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
-                                               // 2-126, 4-124, 6-122, 8-120,
-    const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
-                                               // 3-127, 5-125, 7-123, 9-121,
-    // Calculate product into 'y'.
-    //    yr = wkr * xr - wki * xi;
-    //    yi = wkr * xi + wki * xr;
-    const __m128 a_ = _mm_mul_ps(wkr_, xr_);
-    const __m128 b_ = _mm_mul_ps(wki_, xi_);
-    const __m128 c_ = _mm_mul_ps(wkr_, xi_);
-    const __m128 d_ = _mm_mul_ps(wki_, xr_);
-    const __m128 yr_ = _mm_sub_ps(a_, b_);     // 2-126, 4-124, 6-122, 8-120,
-    const __m128 yi_ = _mm_add_ps(c_, d_);     // 3-127, 5-125, 7-123, 9-121,
-    // Update 'a'.
-    //    a[j2 + 0] -= yr;
-    //    a[j2 + 1] -= yi;
-    //    a[k2 + 0] += yr;
-    //    a[k2 + 1] -= yi;
-    const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_);  //   2,   4,   6,   8,
-    const __m128 a_j2_p1n = _mm_sub_ps(a_j2_p1, yi_);  //   3,   5,   7,   9,
-    const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_);  // 126, 124, 122, 120,
-    const __m128 a_k2_p1n = _mm_sub_ps(a_k2_p1, yi_);  // 127, 125, 123, 121,
-    // Shuffle in right order and store.
-    const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
-                                                       //   2,   3,   4,   5,
-    const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
-                                                       //   6,   7,   8,   9,
-    const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
-                                                       // 122, 123, 120, 121,
-    const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
-                                                       // 126, 127, 124, 125,
-    const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
-                            _MM_SHUFFLE(1, 0, 3 ,2));  // 120, 121, 122, 123,
-    const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
-                            _MM_SHUFFLE(1, 0, 3 ,2));  // 124, 125, 126, 127,
-    _mm_storeu_ps(&a[0   + j2], a_j2_0n);
-    _mm_storeu_ps(&a[4   + j2], a_j2_4n);
-    _mm_storeu_ps(&a[122 - j2], a_k2_0n);
-    _mm_storeu_ps(&a[126 - j2], a_k2_4n);
-  }
-  // Scalar code for the remaining items.
-  for (; j2 < 64; j1 += 1, j2 += 2) {
-    k2 = 128 - j2;
-    k1 =  32 - j1;
-    wkr = 0.5f - c[k1];
-    wki = c[j1];
-    xr = a[j2 + 0] - a[k2 + 0];
-    xi = a[j2 + 1] + a[k2 + 1];
-    yr = wkr * xr - wki * xi;
-    yi = wkr * xi + wki * xr;
-    a[j2 + 0] -= yr;
-    a[j2 + 1] -= yi;
-    a[k2 + 0] += yr;
-    a[k2 + 1] -= yi;
-  }
-}
-
-static void rftbsub_128_SSE2(float *a) {
-  const float *c = rdft_w + 32;
-  int j1, j2, k1, k2;
-  float wkr, wki, xr, xi, yr, yi;
-
-  static const ALIGN16_BEG float ALIGN16_END k_half[4] =
-    {0.5f, 0.5f, 0.5f, 0.5f};
-  const __m128 mm_half = _mm_load_ps(k_half);
-
-  a[1] = -a[1];
-  // Vectorized code (four at once).
-  //    Note: commented number are indexes for the first iteration of the loop.
-  for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
-    // Load 'wk'.
-    const __m128 c_j1 = _mm_loadu_ps(&c[     j1]);         //  1,  2,  3,  4,
-    const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]);         // 28, 29, 30, 31,
-    const __m128 wkrt = _mm_sub_ps(mm_half, c_k1);         // 28, 29, 30, 31,
-    const __m128 wkr_ =
-      _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
-    const __m128 wki_ = c_j1;                              //  1,  2,  3,  4,
-    // Load and shuffle 'a'.
-    const __m128 a_j2_0 = _mm_loadu_ps(&a[0   + j2]);  //   2,   3,   4,   5,
-    const __m128 a_j2_4 = _mm_loadu_ps(&a[4   + j2]);  //   6,   7,   8,   9,
-    const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]);  // 120, 121, 122, 123,
-    const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]);  // 124, 125, 126, 127,
-    const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
-                            _MM_SHUFFLE(2, 0, 2 ,0));  //   2,   4,   6,   8,
-    const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
-                            _MM_SHUFFLE(3, 1, 3 ,1));  //   3,   5,   7,   9,
-    const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
-                            _MM_SHUFFLE(0, 2, 0 ,2));  // 126, 124, 122, 120,
-    const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
-                            _MM_SHUFFLE(1, 3, 1 ,3));  // 127, 125, 123, 121,
-    // Calculate 'x'.
-    const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
-                                               // 2-126, 4-124, 6-122, 8-120,
-    const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
-                                               // 3-127, 5-125, 7-123, 9-121,
-    // Calculate product into 'y'.
-    //    yr = wkr * xr + wki * xi;
-    //    yi = wkr * xi - wki * xr;
-    const __m128 a_ = _mm_mul_ps(wkr_, xr_);
-    const __m128 b_ = _mm_mul_ps(wki_, xi_);
-    const __m128 c_ = _mm_mul_ps(wkr_, xi_);
-    const __m128 d_ = _mm_mul_ps(wki_, xr_);
-    const __m128 yr_ = _mm_add_ps(a_, b_);     // 2-126, 4-124, 6-122, 8-120,
-    const __m128 yi_ = _mm_sub_ps(c_, d_);     // 3-127, 5-125, 7-123, 9-121,
-    // Update 'a'.
-    //    a[j2 + 0] = a[j2 + 0] - yr;
-    //    a[j2 + 1] = yi - a[j2 + 1];
-    //    a[k2 + 0] = yr + a[k2 + 0];
-    //    a[k2 + 1] = yi - a[k2 + 1];
-    const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_);  //   2,   4,   6,   8,
-    const __m128 a_j2_p1n = _mm_sub_ps(yi_, a_j2_p1);  //   3,   5,   7,   9,
-    const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_);  // 126, 124, 122, 120,
-    const __m128 a_k2_p1n = _mm_sub_ps(yi_, a_k2_p1);  // 127, 125, 123, 121,
-    // Shuffle in right order and store.
-    const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
-                                                       //   2,   3,   4,   5,
-    const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
-                                                       //   6,   7,   8,   9,
-    const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
-                                                       // 122, 123, 120, 121,
-    const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
-                                                       // 126, 127, 124, 125,
-    const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
-                            _MM_SHUFFLE(1, 0, 3 ,2));  // 120, 121, 122, 123,
-    const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
-                            _MM_SHUFFLE(1, 0, 3 ,2));  // 124, 125, 126, 127,
-    _mm_storeu_ps(&a[0   + j2], a_j2_0n);
-    _mm_storeu_ps(&a[4   + j2], a_j2_4n);
-    _mm_storeu_ps(&a[122 - j2], a_k2_0n);
-    _mm_storeu_ps(&a[126 - j2], a_k2_4n);
-  }
-  // Scalar code for the remaining items.
-  for (; j2 < 64; j1 += 1, j2 += 2) {
-    k2 = 128 - j2;
-    k1 =  32 - j1;
-    wkr = 0.5f - c[k1];
-    wki = c[j1];
-    xr = a[j2 + 0] - a[k2 + 0];
-    xi = a[j2 + 1] + a[k2 + 1];
-    yr = wkr * xr + wki * xi;
-    yi = wkr * xi - wki * xr;
-    a[j2 + 0] = a[j2 + 0] - yr;
-    a[j2 + 1] = yi - a[j2 + 1];
-    a[k2 + 0] = yr + a[k2 + 0];
-    a[k2 + 1] = yi - a[k2 + 1];
-  }
-  a[65] = -a[65];
-}
-
-void aec_rdft_init_sse2(void) {
-  cft1st_128 = cft1st_128_SSE2;
-  cftmdl_128 = cftmdl_128_SSE2;
-  rftfsub_128 = rftfsub_128_SSE2;
-  rftbsub_128 = rftbsub_128_SSE2;
-}
-
-#endif  // WEBRTC_USE_SS2

src/modules/audio_processing/aec/echo_cancellation.c

@@ -1,901 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Contains the API functions for the AEC.
- */
-#include "echo_cancellation.h"
-
-#include <math.h>
-#ifdef AEC_DEBUG
-#include <stdio.h>
-#endif
-#include <stdlib.h>
-#include <string.h>
-
-#include "aec_core.h"
-#include "resampler.h"
-#include "ring_buffer.h"
-
-#define BUF_SIZE_FRAMES 50 // buffer size (frames)
-// Maximum length of resampled signal. Must be an integer multiple of frames
-// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
-// The factor of 2 handles wb, and the + 1 is as a safety margin
-#define MAX_RESAMP_LEN (5 * FRAME_LEN)
-
-static const int bufSizeSamp = BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
-static const int sampMsNb = 8; // samples per ms in nb
-// Target suppression levels for nlp modes
-// log{0.001, 0.00001, 0.00000001}
-static const float targetSupp[3] = {-6.9f, -11.5f, -18.4f};
-static const float minOverDrive[3] = {1.0f, 2.0f, 5.0f};
-static const int initCheck = 42;
-
-typedef struct {
-    int delayCtr;
-    int sampFreq;
-    int splitSampFreq;
-    int scSampFreq;
-    float sampFactor; // scSampRate / sampFreq
-    short nlpMode;
-    short autoOnOff;
-    short activity;
-    short skewMode;
-    short bufSizeStart;
-    //short bufResetCtr;  // counts number of noncausal frames
-    int knownDelay;
-
-    // Stores the last frame added to the farend buffer
-    short farendOld[2][FRAME_LEN];
-    short initFlag; // indicates if AEC has been initialized
-
-    // Variables used for averaging far end buffer size
-    short counter;
-    short sum;
-    short firstVal;
-    short checkBufSizeCtr;
-
-    // Variables used for delay shifts
-    short msInSndCardBuf;
-    short filtDelay;
-    int timeForDelayChange;
-    int ECstartup;
-    int checkBuffSize;
-    int delayChange;
-    short lastDelayDiff;
-
-#ifdef AEC_DEBUG
-    FILE *bufFile;
-    FILE *delayFile;
-    FILE *skewFile;
-    FILE *preCompFile;
-    FILE *postCompFile;
-#endif // AEC_DEBUG
-
-    // Structures
-    void *farendBuf;
-    void *resampler;
-
-    int skewFrCtr;
-    int resample; // if the skew is small enough we don't resample
-    int highSkewCtr;
-    float skew;
-
-    int lastError;
-
-    aec_t *aec;
-} aecpc_t;
-
-// Estimates delay to set the position of the farend buffer read pointer
-// (controlled by knownDelay)
-static int EstBufDelay(aecpc_t *aecInst, short msInSndCardBuf);
-
-// Stuffs the farend buffer if the estimated delay is too large
-static int DelayComp(aecpc_t *aecInst);
-
-WebRtc_Word32 WebRtcAec_Create(void **aecInst)
-{
-    aecpc_t *aecpc;
-    if (aecInst == NULL) {
-        return -1;
-    }
-
-    aecpc = malloc(sizeof(aecpc_t));
-    *aecInst = aecpc;
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (WebRtcAec_CreateAec(&aecpc->aec) == -1) {
-        WebRtcAec_Free(aecpc);
-        aecpc = NULL;
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aecpc->farendBuf, bufSizeSamp) == -1) {
-        WebRtcAec_Free(aecpc);
-        aecpc = NULL;
-        return -1;
-    }
-
-    if (WebRtcAec_CreateResampler(&aecpc->resampler) == -1) {
-        WebRtcAec_Free(aecpc);
-        aecpc = NULL;
-        return -1;
-    }
-
-    aecpc->initFlag = 0;
-    aecpc->lastError = 0;
-
-#ifdef AEC_DEBUG
-    aecpc->aec->farFile = fopen("aecFar.pcm","wb");
-    aecpc->aec->nearFile = fopen("aecNear.pcm","wb");
-    aecpc->aec->outFile = fopen("aecOut.pcm","wb");
-    aecpc->aec->outLpFile = fopen("aecOutLp.pcm","wb");
-
-    aecpc->bufFile = fopen("aecBuf.dat", "wb");
-    aecpc->skewFile = fopen("aecSkew.dat", "wb");
-    aecpc->delayFile = fopen("aecDelay.dat", "wb");
-    aecpc->preCompFile = fopen("preComp.pcm", "wb");
-    aecpc->postCompFile = fopen("postComp.pcm", "wb");
-#endif // AEC_DEBUG
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_Free(void *aecInst)
-{
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-#ifdef AEC_DEBUG
-    fclose(aecpc->aec->farFile);
-    fclose(aecpc->aec->nearFile);
-    fclose(aecpc->aec->outFile);
-    fclose(aecpc->aec->outLpFile);
-
-    fclose(aecpc->bufFile);
-    fclose(aecpc->skewFile);
-    fclose(aecpc->delayFile);
-    fclose(aecpc->preCompFile);
-    fclose(aecpc->postCompFile);
-#endif // AEC_DEBUG
-
-    WebRtcAec_FreeAec(aecpc->aec);
-    WebRtcApm_FreeBuffer(aecpc->farendBuf);
-    WebRtcAec_FreeResampler(aecpc->resampler);
-    free(aecpc);
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_Init(void *aecInst, WebRtc_Word32 sampFreq, WebRtc_Word32 scSampFreq)
-{
-    aecpc_t *aecpc = aecInst;
-    AecConfig aecConfig;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (sampFreq != 8000 && sampFreq != 16000  && sampFreq != 32000) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecpc->sampFreq = sampFreq;
-
-    if (scSampFreq < 1 || scSampFreq > 96000) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecpc->scSampFreq = scSampFreq;
-
-    // Initialize echo canceller core
-    if (WebRtcAec_InitAec(aecpc->aec, aecpc->sampFreq) == -1) {
-        aecpc->lastError = AEC_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    // Initialize farend buffer
-    if (WebRtcApm_InitBuffer(aecpc->farendBuf) == -1) {
-        aecpc->lastError = AEC_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) {
-        aecpc->lastError = AEC_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    aecpc->initFlag = initCheck;  // indicates that initialization has been done
-
-    if (aecpc->sampFreq == 32000) {
-        aecpc->splitSampFreq = 16000;
-    }
-    else {
-        aecpc->splitSampFreq = sampFreq;
-    }
-
-    aecpc->skewFrCtr = 0;
-    aecpc->activity = 0;
-
-    aecpc->delayChange = 1;
-    aecpc->delayCtr = 0;
-
-    aecpc->sum = 0;
-    aecpc->counter = 0;
-    aecpc->checkBuffSize = 1;
-    aecpc->firstVal = 0;
-
-    aecpc->ECstartup = 1;
-    aecpc->bufSizeStart = 0;
-    aecpc->checkBufSizeCtr = 0;
-    aecpc->filtDelay = 0;
-    aecpc->timeForDelayChange =0;
-    aecpc->knownDelay = 0;
-    aecpc->lastDelayDiff = 0;
-
-    aecpc->skew = 0;
-    aecpc->resample = kAecFalse;
-    aecpc->highSkewCtr = 0;
-    aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq;
-
-    memset(&aecpc->farendOld[0][0], 0, 160);
-
-    // Default settings.
-    aecConfig.nlpMode = kAecNlpModerate;
-    aecConfig.skewMode = kAecFalse;
-    aecConfig.metricsMode = kAecFalse;
-    aecConfig.delay_logging = kAecFalse;
-
-    if (WebRtcAec_set_config(aecpc, aecConfig) == -1) {
-        aecpc->lastError = AEC_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    return 0;
-}
-
-// only buffer L band for farend
-WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst, const WebRtc_Word16 *farend,
-    WebRtc_Word16 nrOfSamples)
-{
-    aecpc_t *aecpc = aecInst;
-    WebRtc_Word32 retVal = 0;
-    short newNrOfSamples;
-    short newFarend[MAX_RESAMP_LEN];
-    float skew;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (farend == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    // number of samples == 160 for SWB input
-    if (nrOfSamples != 80 && nrOfSamples != 160) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-
-    skew = aecpc->skew;
-
-    // TODO: Is this really a good idea?
-    if (!aecpc->ECstartup) {
-        DelayComp(aecpc);
-    }
-
-    if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
-        // Resample and get a new number of samples
-        newNrOfSamples = WebRtcAec_ResampleLinear(aecpc->resampler,
-                                                  farend,
-                                                  nrOfSamples,
-                                                  skew,
-                                                  newFarend);
-        WebRtcApm_WriteBuffer(aecpc->farendBuf, newFarend, newNrOfSamples);
-
-#ifdef AEC_DEBUG
-        fwrite(farend, 2, nrOfSamples, aecpc->preCompFile);
-        fwrite(newFarend, 2, newNrOfSamples, aecpc->postCompFile);
-#endif
-    }
-    else {
-        WebRtcApm_WriteBuffer(aecpc->farendBuf, farend, nrOfSamples);
-    }
-
-    return retVal;
-}
-
-WebRtc_Word32 WebRtcAec_Process(void *aecInst, const WebRtc_Word16 *nearend,
-    const WebRtc_Word16 *nearendH, WebRtc_Word16 *out, WebRtc_Word16 *outH,
-    WebRtc_Word16 nrOfSamples, WebRtc_Word16 msInSndCardBuf, WebRtc_Word32 skew)
-{
-    aecpc_t *aecpc = aecInst;
-    WebRtc_Word32 retVal = 0;
-    short i;
-    short farend[FRAME_LEN];
-    short nmbrOfFilledBuffers;
-    short nBlocks10ms;
-    short nFrames;
-#ifdef AEC_DEBUG
-    short msInAECBuf;
-#endif
-    // Limit resampling to doubling/halving of signal
-    const float minSkewEst = -0.5f;
-    const float maxSkewEst = 1.0f;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (nearend == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (out == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    // number of samples == 160 for SWB input
-    if (nrOfSamples != 80 && nrOfSamples != 160) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-
-    // Check for valid pointers based on sampling rate
-    if (aecpc->sampFreq == 32000 && nearendH == NULL) {
-       aecpc->lastError = AEC_NULL_POINTER_ERROR;
-       return -1;
-    }
-
-    if (msInSndCardBuf < 0) {
-        msInSndCardBuf = 0;
-        aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
-        retVal = -1;
-    }
-    else if (msInSndCardBuf > 500) {
-        msInSndCardBuf = 500;
-        aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
-        retVal = -1;
-    }
-    msInSndCardBuf += 10;
-    aecpc->msInSndCardBuf = msInSndCardBuf;
-
-    if (aecpc->skewMode == kAecTrue) {
-        if (aecpc->skewFrCtr < 25) {
-            aecpc->skewFrCtr++;
-        }
-        else {
-            retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
-            if (retVal == -1) {
-                aecpc->skew = 0;
-                aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
-            }
-
-            aecpc->skew /= aecpc->sampFactor*nrOfSamples;
-
-            if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) {
-                aecpc->resample = kAecFalse;
-            }
-            else {
-                aecpc->resample = kAecTrue;
-            }
-
-            if (aecpc->skew < minSkewEst) {
-                aecpc->skew = minSkewEst;
-            }
-            else if (aecpc->skew > maxSkewEst) {
-                aecpc->skew = maxSkewEst;
-            }
-
-#ifdef AEC_DEBUG
-            fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
-#endif
-        }
-    }
-
-    nFrames = nrOfSamples / FRAME_LEN;
-    nBlocks10ms = nFrames / aecpc->aec->mult;
-
-    if (aecpc->ECstartup) {
-        if (nearend != out) {
-            // Only needed if they don't already point to the same place.
-            memcpy(out, nearend, sizeof(short) * nrOfSamples);
-        }
-        nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecpc->farendBuf) / FRAME_LEN;
-
-        // The AEC is in the start up mode
-        // AEC is disabled until the soundcard buffer and farend buffers are OK
-
-        // Mechanism to ensure that the soundcard buffer is reasonably stable.
-        if (aecpc->checkBuffSize) {
-
-            aecpc->checkBufSizeCtr++;
-            // Before we fill up the far end buffer we require the amount of data on the
-            // sound card to be stable (+/-8 ms) compared to the first value. This
-            // comparison is made during the following 4 consecutive frames. If it seems
-            // to be stable then we start to fill up the far end buffer.
-
-            if (aecpc->counter == 0) {
-                aecpc->firstVal = aecpc->msInSndCardBuf;
-                aecpc->sum = 0;
-            }
-
-            if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) <
-                WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) {
-                aecpc->sum += aecpc->msInSndCardBuf;
-                aecpc->counter++;
-            }
-            else {
-                aecpc->counter = 0;
-            }
-
-            if (aecpc->counter*nBlocks10ms >= 6) {
-                // The farend buffer size is determined in blocks of 80 samples
-                // Use 75% of the average value of the soundcard buffer
-                aecpc->bufSizeStart = WEBRTC_SPL_MIN((int) (0.75 * (aecpc->sum *
-                    aecpc->aec->mult) / (aecpc->counter * 10)), BUF_SIZE_FRAMES);
-                // buffersize has now been determined
-                aecpc->checkBuffSize = 0;
-            }
-
-            if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) {
-                // for really bad sound cards, don't disable echocanceller for more than 0.5 sec
-                aecpc->bufSizeStart = WEBRTC_SPL_MIN((int) (0.75 * (aecpc->msInSndCardBuf *
-                    aecpc->aec->mult) / 10), BUF_SIZE_FRAMES);
-                aecpc->checkBuffSize = 0;
-            }
-        }
-
-        // if checkBuffSize changed in the if-statement above
-        if (!aecpc->checkBuffSize) {
-            // soundcard buffer is now reasonably stable
-            // When the far end buffer is filled with approximately the same amount of
-            // data as the amount on the sound card we end the start up phase and start
-            // to cancel echoes.
-
-            if (nmbrOfFilledBuffers == aecpc->bufSizeStart) {
-                aecpc->ECstartup = 0;  // Enable the AEC
-            }
-            else if (nmbrOfFilledBuffers > aecpc->bufSizeStart) {
-                WebRtcApm_FlushBuffer(aecpc->farendBuf, WebRtcApm_get_buffer_size(aecpc->farendBuf) -
-                    aecpc->bufSizeStart * FRAME_LEN);
-                aecpc->ECstartup = 0;
-            }
-        }
-
-    }
-    else {
-        // AEC is enabled
-
-        // Note only 1 block supported for nb and 2 blocks for wb
-        for (i = 0; i < nFrames; i++) {
-            nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecpc->farendBuf) / FRAME_LEN;
-
-            // Check that there is data in the far end buffer
-            if (nmbrOfFilledBuffers > 0) {
-                // Get the next 80 samples from the farend buffer
-                WebRtcApm_ReadBuffer(aecpc->farendBuf, farend, FRAME_LEN);
-
-                // Always store the last frame for use when we run out of data
-                memcpy(&(aecpc->farendOld[i][0]), farend, FRAME_LEN * sizeof(short));
-            }
-            else {
-                // We have no data so we use the last played frame
-                memcpy(farend, &(aecpc->farendOld[i][0]), FRAME_LEN * sizeof(short));
-            }
-
-            // Call buffer delay estimator when all data is extracted,
-            // i.e. i = 0 for NB and i = 1 for WB or SWB
-            if ((i == 0 && aecpc->splitSampFreq == 8000) ||
-                    (i == 1 && (aecpc->splitSampFreq == 16000))) {
-                EstBufDelay(aecpc, aecpc->msInSndCardBuf);
-            }
-
-            // Call the AEC
-           WebRtcAec_ProcessFrame(aecpc->aec, farend, &nearend[FRAME_LEN * i], &nearendH[FRAME_LEN * i],
-               &out[FRAME_LEN * i], &outH[FRAME_LEN * i], aecpc->knownDelay);
-        }
-    }
-
-#ifdef AEC_DEBUG
-    msInAECBuf = WebRtcApm_get_buffer_size(aecpc->farendBuf) / (sampMsNb*aecpc->aec->mult);
-    fwrite(&msInAECBuf, 2, 1, aecpc->bufFile);
-    fwrite(&(aecpc->knownDelay), sizeof(aecpc->knownDelay), 1, aecpc->delayFile);
-#endif
-
-    return retVal;
-}
-
-WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config)
-{
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    if (config.skewMode != kAecFalse && config.skewMode != kAecTrue) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecpc->skewMode = config.skewMode;
-
-    if (config.nlpMode != kAecNlpConservative && config.nlpMode !=
-            kAecNlpModerate && config.nlpMode != kAecNlpAggressive) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecpc->nlpMode = config.nlpMode;
-    aecpc->aec->targetSupp = targetSupp[aecpc->nlpMode];
-    aecpc->aec->minOverDrive = minOverDrive[aecpc->nlpMode];
-
-    if (config.metricsMode != kAecFalse && config.metricsMode != kAecTrue) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecpc->aec->metricsMode = config.metricsMode;
-    if (aecpc->aec->metricsMode == kAecTrue) {
-        WebRtcAec_InitMetrics(aecpc->aec);
-    }
-
-  if (config.delay_logging != kAecFalse && config.delay_logging != kAecTrue) {
-    aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-    return -1;
-  }
-  aecpc->aec->delay_logging_enabled = config.delay_logging;
-  if (aecpc->aec->delay_logging_enabled == kAecTrue) {
-    memset(aecpc->aec->delay_histogram, 0, sizeof(aecpc->aec->delay_histogram));
-  }
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config)
-{
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (config == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    config->nlpMode = aecpc->nlpMode;
-    config->skewMode = aecpc->skewMode;
-    config->metricsMode = aecpc->aec->metricsMode;
-    config->delay_logging = aecpc->aec->delay_logging_enabled;
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status)
-{
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (status == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    *status = aecpc->aec->echoState;
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics)
-{
-    const float upweight = 0.7f;
-    float dtmp;
-    short stmp;
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (metrics == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    // ERL
-    metrics->erl.instant = (short) aecpc->aec->erl.instant;
-
-    if ((aecpc->aec->erl.himean > offsetLevel) && (aecpc->aec->erl.average > offsetLevel)) {
-    // Use a mix between regular average and upper part average
-        dtmp = upweight * aecpc->aec->erl.himean + (1 - upweight) * aecpc->aec->erl.average;
-        metrics->erl.average = (short) dtmp;
-    }
-    else {
-        metrics->erl.average = offsetLevel;
-    }
-
-    metrics->erl.max = (short) aecpc->aec->erl.max;
-
-    if (aecpc->aec->erl.min < (offsetLevel * (-1))) {
-        metrics->erl.min = (short) aecpc->aec->erl.min;
-    }
-    else {
-        metrics->erl.min = offsetLevel;
-    }
-
-    // ERLE
-    metrics->erle.instant = (short) aecpc->aec->erle.instant;
-
-    if ((aecpc->aec->erle.himean > offsetLevel) && (aecpc->aec->erle.average > offsetLevel)) {
-        // Use a mix between regular average and upper part average
-        dtmp =  upweight * aecpc->aec->erle.himean + (1 - upweight) * aecpc->aec->erle.average;
-        metrics->erle.average = (short) dtmp;
-    }
-    else {
-        metrics->erle.average = offsetLevel;
-    }
-
-    metrics->erle.max = (short) aecpc->aec->erle.max;
-
-    if (aecpc->aec->erle.min < (offsetLevel * (-1))) {
-        metrics->erle.min = (short) aecpc->aec->erle.min;
-    } else {
-        metrics->erle.min = offsetLevel;
-    }
-
-    // RERL
-    if ((metrics->erl.average > offsetLevel) && (metrics->erle.average > offsetLevel)) {
-        stmp = metrics->erl.average + metrics->erle.average;
-    }
-    else {
-        stmp = offsetLevel;
-    }
-    metrics->rerl.average = stmp;
-
-    // No other statistics needed, but returned for completeness
-    metrics->rerl.instant = stmp;
-    metrics->rerl.max = stmp;
-    metrics->rerl.min = stmp;
-
-    // A_NLP
-    metrics->aNlp.instant = (short) aecpc->aec->aNlp.instant;
-
-    if ((aecpc->aec->aNlp.himean > offsetLevel) && (aecpc->aec->aNlp.average > offsetLevel)) {
-        // Use a mix between regular average and upper part average
-        dtmp =  upweight * aecpc->aec->aNlp.himean + (1 - upweight) * aecpc->aec->aNlp.average;
-        metrics->aNlp.average = (short) dtmp;
-    }
-    else {
-        metrics->aNlp.average = offsetLevel;
-    }
-
-    metrics->aNlp.max = (short) aecpc->aec->aNlp.max;
-
-    if (aecpc->aec->aNlp.min < (offsetLevel * (-1))) {
-        metrics->aNlp.min = (short) aecpc->aec->aNlp.min;
-    }
-    else {
-        metrics->aNlp.min = offsetLevel;
-    }
-
-    return 0;
-}
-
-int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std) {
-  aecpc_t* self = handle;
-  int i = 0;
-  int delay_values = 0;
-  int num_delay_values = 0;
-  int my_median = 0;
-  const int kMsPerBlock = (PART_LEN * 1000) / self->splitSampFreq;
-  float l1_norm = 0;
-
-  if (self == NULL) {
-    return -1;
-  }
-  if (median == NULL) {
-    self->lastError = AEC_NULL_POINTER_ERROR;
-    return -1;
-  }
-  if (std == NULL) {
-    self->lastError = AEC_NULL_POINTER_ERROR;
-    return -1;
-  }
-  if (self->initFlag != initCheck) {
-    self->lastError = AEC_UNINITIALIZED_ERROR;
-    return -1;
-  }
-  if (self->aec->delay_logging_enabled == 0) {
-    // Logging disabled
-    self->lastError = AEC_UNSUPPORTED_FUNCTION_ERROR;
-    return -1;
-  }
-
-  // Get number of delay values since last update
-  for (i = 0; i < kMaxDelay; i++) {
-    num_delay_values += self->aec->delay_histogram[i];
-  }
-  if (num_delay_values == 0) {
-    // We have no new delay value data
-    *median = -1;
-    *std = -1;
-    return 0;
-  }
-
-  delay_values = num_delay_values >> 1; // Start value for median count down
-  // Get median of delay values since last update
-  for (i = 0; i < kMaxDelay; i++) {
-    delay_values -= self->aec->delay_histogram[i];
-    if (delay_values < 0) {
-      my_median = i;
-      break;
-    }
-  }
-  *median = my_median * kMsPerBlock;
-
-  // Calculate the L1 norm, with median value as central moment
-  for (i = 0; i < kMaxDelay; i++) {
-    l1_norm += (float) (fabs(i - my_median) * self->aec->delay_histogram[i]);
-  }
-  *std = (int) (l1_norm / (float) num_delay_values + 0.5f) * kMsPerBlock;
-
-  // Reset histogram
-  memset(self->aec->delay_histogram, 0, sizeof(self->aec->delay_histogram));
-
-  return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len)
-{
-    const char version[] = "AEC 2.5.0";
-    const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
-
-    if (versionStr == NULL) {
-        return -1;
-    }
-
-    if (versionLen > len) {
-        return -1;
-    }
-
-    strncpy(versionStr, version, versionLen);
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst)
-{
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    return aecpc->lastError;
-}
-
-static int EstBufDelay(aecpc_t *aecpc, short msInSndCardBuf)
-{
-    short delayNew, nSampFar, nSampSndCard;
-    short diff;
-
-    nSampFar = WebRtcApm_get_buffer_size(aecpc->farendBuf);
-    nSampSndCard = msInSndCardBuf * sampMsNb * aecpc->aec->mult;
-
-    delayNew = nSampSndCard - nSampFar;
-
-    // Account for resampling frame delay
-    if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
-        delayNew -= kResamplingDelay;
-    }
-
-    if (delayNew < FRAME_LEN) {
-        WebRtcApm_FlushBuffer(aecpc->farendBuf, FRAME_LEN);
-        delayNew += FRAME_LEN;
-    }
-
-    aecpc->filtDelay = WEBRTC_SPL_MAX(0, (short)(0.8*aecpc->filtDelay + 0.2*delayNew));
-
-    diff = aecpc->filtDelay - aecpc->knownDelay;
-    if (diff > 224) {
-        if (aecpc->lastDelayDiff < 96) {
-            aecpc->timeForDelayChange = 0;
-        }
-        else {
-            aecpc->timeForDelayChange++;
-        }
-    }
-    else if (diff < 96 && aecpc->knownDelay > 0) {
-        if (aecpc->lastDelayDiff > 224) {
-            aecpc->timeForDelayChange = 0;
-        }
-        else {
-            aecpc->timeForDelayChange++;
-        }
-    }
-    else {
-        aecpc->timeForDelayChange = 0;
-    }
-    aecpc->lastDelayDiff = diff;
-
-    if (aecpc->timeForDelayChange > 25) {
-        aecpc->knownDelay = WEBRTC_SPL_MAX((int)aecpc->filtDelay - 160, 0);
-    }
-    return 0;
-}
-
-static int DelayComp(aecpc_t *aecpc)
-{
-    int nSampFar, nSampSndCard, delayNew, nSampAdd;
-    const int maxStuffSamp = 10 * FRAME_LEN;
-
-    nSampFar = WebRtcApm_get_buffer_size(aecpc->farendBuf);
-    nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->aec->mult;
-    delayNew = nSampSndCard - nSampFar;
-
-    // Account for resampling frame delay
-    if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
-        delayNew -= kResamplingDelay;
-    }
-
-    if (delayNew > FAR_BUF_LEN - FRAME_LEN*aecpc->aec->mult) {
-        // The difference of the buffersizes is larger than the maximum
-        // allowed known delay. Compensate by stuffing the buffer.
-        nSampAdd = (int)(WEBRTC_SPL_MAX((int)(0.5 * nSampSndCard - nSampFar),
-                    FRAME_LEN));
-        nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
-
-        WebRtcApm_StuffBuffer(aecpc->farendBuf, nSampAdd);
-        aecpc->delayChange = 1; // the delay needs to be updated
-    }
-
-    return 0;
-}

src/modules/audio_processing/aec/interface/echo_cancellation.h

@@ -1,278 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_
-
-#include "typedefs.h"
-
-// Errors
-#define AEC_UNSPECIFIED_ERROR           12000
-#define AEC_UNSUPPORTED_FUNCTION_ERROR  12001
-#define AEC_UNINITIALIZED_ERROR         12002
-#define AEC_NULL_POINTER_ERROR          12003
-#define AEC_BAD_PARAMETER_ERROR         12004
-
-// Warnings
-#define AEC_BAD_PARAMETER_WARNING       12050
-
-enum {
-    kAecNlpConservative = 0,
-    kAecNlpModerate,
-    kAecNlpAggressive
-};
-
-enum {
-    kAecFalse = 0,
-    kAecTrue
-};
-
-typedef struct {
-    WebRtc_Word16 nlpMode;        // default kAecNlpModerate
-    WebRtc_Word16 skewMode;       // default kAecFalse
-    WebRtc_Word16 metricsMode;    // default kAecFalse
-    int delay_logging;            // default kAecFalse
-    //float realSkew;
-} AecConfig;
-
-typedef struct {
-    WebRtc_Word16 instant;
-    WebRtc_Word16 average;
-    WebRtc_Word16 max;
-    WebRtc_Word16 min;
-} AecLevel;
-
-typedef struct {
-    AecLevel rerl;
-    AecLevel erl;
-    AecLevel erle;
-    AecLevel aNlp;
-} AecMetrics;
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * Allocates the memory needed by the AEC. The memory needs to be initialized
- * separately using the WebRtcAec_Init() function.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void **aecInst               Pointer to the AEC instance to be created
- *                              and initialized
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return          0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_Create(void **aecInst);
-
-/*
- * This function releases the memory allocated by WebRtcAec_Create().
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void         *aecInst        Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_Free(void *aecInst);
-
-/*
- * Initializes an AEC instance.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- * WebRtc_Word32  sampFreq      Sampling frequency of data
- * WebRtc_Word32  scSampFreq    Soundcard sampling frequency
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return          0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_Init(void *aecInst,
-                             WebRtc_Word32 sampFreq,
-                             WebRtc_Word32 scSampFreq);
-
-/*
- * Inserts an 80 or 160 sample block of data into the farend buffer.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- * WebRtc_Word16  *farend       In buffer containing one frame of
- *                              farend signal for L band
- * WebRtc_Word16  nrOfSamples   Number of samples in farend buffer
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst,
-                                     const WebRtc_Word16 *farend,
-                                     WebRtc_Word16 nrOfSamples);
-
-/*
- * Runs the echo canceller on an 80 or 160 sample blocks of data.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void          *aecInst       Pointer to the AEC instance
- * WebRtc_Word16 *nearend       In buffer containing one frame of
- *                              nearend+echo signal for L band
- * WebRtc_Word16 *nearendH      In buffer containing one frame of
- *                              nearend+echo signal for H band
- * WebRtc_Word16 nrOfSamples    Number of samples in nearend buffer
- * WebRtc_Word16 msInSndCardBuf Delay estimate for sound card and
- *                              system buffers
- * WebRtc_Word16 skew           Difference between number of samples played
- *                              and recorded at the soundcard (for clock skew
- *                              compensation)
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word16  *out          Out buffer, one frame of processed nearend
- *                              for L band
- * WebRtc_Word16  *outH         Out buffer, one frame of processed nearend
- *                              for H band
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_Process(void *aecInst,
-                                const WebRtc_Word16 *nearend,
-                                const WebRtc_Word16 *nearendH,
-                                WebRtc_Word16 *out,
-                                WebRtc_Word16 *outH,
-                                WebRtc_Word16 nrOfSamples,
-                                WebRtc_Word16 msInSndCardBuf,
-                                WebRtc_Word32 skew);
-
-/*
- * This function enables the user to set certain parameters on-the-fly.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- * AecConfig      config        Config instance that contains all
- *                              properties to be set
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config);
-
-/*
- * Gets the on-the-fly paramters.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * AecConfig      *config       Pointer to the config instance that
- *                              all properties will be written to
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config);
-
-/*
- * Gets the current echo status of the nearend signal.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word16  *status       0: Almost certainly nearend single-talk
- *                              1: Might not be neared single-talk
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status);
-
-/*
- * Gets the current echo metrics for the session.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * AecMetrics     *metrics      Struct which will be filled out with the
- *                              current echo metrics.
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics);
-
-/*
- * Gets the current delay metrics for the session.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void*      handle            Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * int*       median            Delay median value.
- * int*       std               Delay standard deviation.
- *
- * int        return             0: OK
- *                              -1: error
- */
-int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std);
-
-/*
- * Gets the last error code.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        11000-11100: error code
- */
-WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst);
-
-/*
- * Gets a version string.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * char           *versionStr   Pointer to a string array
- * WebRtc_Word16  len           The maximum length of the string
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word8   *versionStr   Pointer to a string array
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len);
-
-#ifdef __cplusplus
-}
-#endif
-#endif  /* WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_ */

src/modules/audio_processing/aec/resampler.c

@@ -1,233 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/* Resamples a signal to an arbitrary rate. Used by the AEC to compensate for clock
- * skew by resampling the farend signal.
- */
-
-#include <assert.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-
-#include "resampler.h"
-#include "aec_core.h"
-
-enum { kFrameBufferSize = FRAME_LEN * 4 };
-enum { kEstimateLengthFrames = 400 };
-
-typedef struct {
-    short buffer[kFrameBufferSize];
-    float position;
-
-    int deviceSampleRateHz;
-    int skewData[kEstimateLengthFrames];
-    int skewDataIndex;
-    float skewEstimate;
-} resampler_t;
-
-static int EstimateSkew(const int* rawSkew,
-                        int size,
-                        int absLimit,
-                        float *skewEst);
-
-int WebRtcAec_CreateResampler(void **resampInst)
-{
-    resampler_t *obj = malloc(sizeof(resampler_t));
-    *resampInst = obj;
-    if (obj == NULL) {
-        return -1;
-    }
-
-    return 0;
-}
-
-int WebRtcAec_InitResampler(void *resampInst, int deviceSampleRateHz)
-{
-    resampler_t *obj = (resampler_t*) resampInst;
-    memset(obj->buffer, 0, sizeof(obj->buffer));
-    obj->position = 0.0;
-
-    obj->deviceSampleRateHz = deviceSampleRateHz;
-    memset(obj->skewData, 0, sizeof(obj->skewData));
-    obj->skewDataIndex = 0;
-    obj->skewEstimate = 0.0;
-
-    return 0;
-}
-
-int WebRtcAec_FreeResampler(void *resampInst)
-{
-    resampler_t *obj = (resampler_t*) resampInst;
-    free(obj);
-
-    return 0;
-}
-
-int WebRtcAec_ResampleLinear(void *resampInst,
-                             const short *inspeech,
-                             int size,
-                             float skew,
-                             short *outspeech)
-{
-    resampler_t *obj = (resampler_t*) resampInst;
-
-    short *y;
-    float be, tnew, interp;
-    int tn, outsize, mm;
-
-    if (size < 0 || size > 2 * FRAME_LEN) {
-        return -1;
-    }
-
-    // Add new frame data in lookahead
-    memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay],
-           inspeech,
-           size * sizeof(short));
-
-    // Sample rate ratio
-    be = 1 + skew;
-
-    // Loop over input frame
-    mm = 0;
-    y = &obj->buffer[FRAME_LEN]; // Point at current frame
-
-    tnew = be * mm + obj->position;
-    tn = (int) tnew;
-
-    while (tn < size) {
-
-        // Interpolation
-        interp = y[tn] + (tnew - tn) * (y[tn+1] - y[tn]);
-
-        if (interp > 32767) {
-            interp = 32767;
-        }
-        else if (interp < -32768) {
-            interp = -32768;
-        }
-
-        outspeech[mm] = (short) interp;
-        mm++;
-
-        tnew = be * mm + obj->position;
-        tn = (int) tnew;
-    }
-
-    outsize = mm;
-    obj->position += outsize * be - size;
-
-    // Shift buffer
-    memmove(obj->buffer,
-            &obj->buffer[size],
-            (kFrameBufferSize - size) * sizeof(short));
-
-    return outsize;
-}
-
-int WebRtcAec_GetSkew(void *resampInst, int rawSkew, float *skewEst)
-{
-    resampler_t *obj = (resampler_t*)resampInst;
-    int err = 0;
-
-    if (obj->skewDataIndex < kEstimateLengthFrames) {
-        obj->skewData[obj->skewDataIndex] = rawSkew;
-        obj->skewDataIndex++;
-    }
-    else if (obj->skewDataIndex == kEstimateLengthFrames) {
-        err = EstimateSkew(obj->skewData,
-                           kEstimateLengthFrames,
-                           obj->deviceSampleRateHz,
-                           skewEst);
-        obj->skewEstimate = *skewEst;
-        obj->skewDataIndex++;
-    }
-    else {
-        *skewEst = obj->skewEstimate;
-    }
-
-    return err;
-}
-
-int EstimateSkew(const int* rawSkew,
-                 int size,
-                 int deviceSampleRateHz,
-                 float *skewEst)
-{
-    const int absLimitOuter = (int)(0.04f * deviceSampleRateHz);
-    const int absLimitInner = (int)(0.0025f * deviceSampleRateHz);
-    int i = 0;
-    int n = 0;
-    float rawAvg = 0;
-    float err = 0;
-    float rawAbsDev = 0;
-    int upperLimit = 0;
-    int lowerLimit = 0;
-    float cumSum = 0;
-    float x = 0;
-    float x2 = 0;
-    float y = 0;
-    float xy = 0;
-    float xAvg = 0;
-    float denom = 0;
-    float skew = 0;
-
-    *skewEst = 0; // Set in case of error below.
-    for (i = 0; i < size; i++) {
-      if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
-        n++;
-        rawAvg += rawSkew[i];
-      }
-    }
-
-    if (n == 0) {
-      return -1;
-    }
-    assert(n > 0);
-    rawAvg /= n;
-
-    for (i = 0; i < size; i++) {
-      if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
-        err = rawSkew[i] - rawAvg;
-        rawAbsDev += err >= 0 ? err : -err;
-      }
-    }
-    assert(n > 0);
-    rawAbsDev /= n;
-    upperLimit = (int)(rawAvg + 5 * rawAbsDev + 1); // +1 for ceiling.
-    lowerLimit = (int)(rawAvg - 5 * rawAbsDev - 1); // -1 for floor.
-
-    n = 0;
-    for (i = 0; i < size; i++) {
-        if ((rawSkew[i] < absLimitInner && rawSkew[i] > -absLimitInner) ||
-            (rawSkew[i] < upperLimit && rawSkew[i] > lowerLimit)) {
-            n++;
-            cumSum += rawSkew[i];
-            x += n;
-            x2 += n*n;
-            y += cumSum;
-            xy += n * cumSum;
-        }
-    }
-
-    if (n == 0) {
-      return -1;
-    }
-    assert(n > 0);
-    xAvg = x / n;
-    denom = x2 - xAvg*x;
-
-    if (denom != 0) {
-        skew = (xy - xAvg*y) / denom;
-    }
-
-    *skewEst = skew;
-    return 0;
-}

src/modules/audio_processing/aec/resampler.h

@@ -1,32 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_RESAMPLER_H_
-
-enum { kResamplingDelay = 1 };
-
-// Unless otherwise specified, functions return 0 on success and -1 on error
-int WebRtcAec_CreateResampler(void **resampInst);
-int WebRtcAec_InitResampler(void *resampInst, int deviceSampleRateHz);
-int WebRtcAec_FreeResampler(void *resampInst);
-
-// Estimates skew from raw measurement.
-int WebRtcAec_GetSkew(void *resampInst, int rawSkew, float *skewEst);
-
-// Resamples input using linear interpolation.
-// Returns size of resampled array.
-int WebRtcAec_ResampleLinear(void *resampInst,
-                             const short *inspeech,
-                             int size,
-                             float skew,
-                             short *outspeech);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_RESAMPLER_H_

src/modules/audio_processing/aecm/Makefile.am

@@ -1,9 +0,0 @@
-noinst_LTLIBRARIES = libaecm.la
-
-libaecm_la_SOURCES = interface/echo_control_mobile.h \
-		     echo_control_mobile.c \
-		     aecm_core.c \
-		     aecm_core.h
-libaecm_la_CFLAGS = $(AM_CFLAGS) $(COMMON_CFLAGS) \
-		    -I$(top_srcdir)/src/common_audio/signal_processing_library/main/interface \
-		    -I$(top_srcdir)/src/modules/audio_processing/utility

src/modules/audio_processing/aecm/aecm.gypi

@@ -1,34 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'targets': [
-    {
-      'target_name': 'aecm',
-      'type': '<(library)',
-      'dependencies': [
-        '<(webrtc_root)/common_audio/common_audio.gyp:spl',
-        'apm_util'
-      ],
-      'include_dirs': [
-        'interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'interface',
-        ],
-      },
-      'sources': [
-        'interface/echo_control_mobile.h',
-        'echo_control_mobile.c',
-        'aecm_core.c',
-        'aecm_core.h',
-      ],
-    },
-  ],
-}

src/modules/audio_processing/aecm/aecm_core.h

@@ -1,358 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Performs echo control (suppression) with fft routines in fixed-point
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_SOURCE_AECM_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_SOURCE_AECM_CORE_H_
-
-#define AECM_DYNAMIC_Q // turn on/off dynamic Q-domain
-//#define AECM_WITH_ABS_APPROX
-//#define AECM_SHORT                // for 32 sample partition length (otherwise 64)
-
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-// Algorithm parameters
-
-#define FRAME_LEN       80              // Total frame length, 10 ms
-#ifdef AECM_SHORT
-
-#define PART_LEN        32              // Length of partition
-#define PART_LEN_SHIFT  6               // Length of (PART_LEN * 2) in base 2
-
-#else
-
-#define PART_LEN        64              // Length of partition
-#define PART_LEN_SHIFT  7               // Length of (PART_LEN * 2) in base 2
-
-#endif
-
-#define PART_LEN1       (PART_LEN + 1)  // Unique fft coefficients
-#define PART_LEN2       (PART_LEN << 1) // Length of partition * 2
-#define PART_LEN4       (PART_LEN << 2) // Length of partition * 4
-#define FAR_BUF_LEN     PART_LEN4       // Length of buffers
-#define MAX_DELAY 100
-
-// Counter parameters
-#ifdef AECM_SHORT
-
-#define CONV_LEN        1024            // Convergence length used at startup
-#else
-
-#define CONV_LEN        512             // Convergence length used at startup
-#endif
-
-#define CONV_LEN2       (CONV_LEN << 1) // Convergence length * 2 used at startup
-// Energy parameters
-#define MAX_BUF_LEN     64              // History length of energy signals
-
-#define FAR_ENERGY_MIN  1025            // Lowest Far energy level: At least 2 in energy
-#define FAR_ENERGY_DIFF 929             // Allowed difference between max and min
-
-#define ENERGY_DEV_OFFSET       0       // The energy error offset in Q8
-#define ENERGY_DEV_TOL  400             // The energy estimation tolerance in Q8
-#define FAR_ENERGY_VAD_REGION   230     // Far VAD tolerance region
-// Stepsize parameters
-#define MU_MIN          10              // Min stepsize 2^-MU_MIN (far end energy dependent)
-#define MU_MAX          1               // Max stepsize 2^-MU_MAX (far end energy dependent)
-#define MU_DIFF         9               // MU_MIN - MU_MAX
-// Channel parameters
-#define MIN_MSE_COUNT   20              // Min number of consecutive blocks with enough far end
-                                        // energy to compare channel estimates
-#define MIN_MSE_DIFF    29              // The ratio between adapted and stored channel to
-                                        // accept a new storage (0.8 in Q-MSE_RESOLUTION)
-#define MSE_RESOLUTION  5               // MSE parameter resolution
-#define RESOLUTION_CHANNEL16    12      // W16 Channel in Q-RESOLUTION_CHANNEL16
-#define RESOLUTION_CHANNEL32    28      // W32 Channel in Q-RESOLUTION_CHANNEL
-#define CHANNEL_VAD     16              // Minimum energy in frequency band to update channel
-// Suppression gain parameters: SUPGAIN_ parameters in Q-(RESOLUTION_SUPGAIN)
-#define RESOLUTION_SUPGAIN      8       // Channel in Q-(RESOLUTION_SUPGAIN)
-#define SUPGAIN_DEFAULT (1 << RESOLUTION_SUPGAIN)   // Default suppression gain
-#define SUPGAIN_ERROR_PARAM_A   3072    // Estimation error parameter (Maximum gain) (8 in Q8)
-#define SUPGAIN_ERROR_PARAM_B   1536    // Estimation error parameter (Gain before going down)
-#define SUPGAIN_ERROR_PARAM_D   SUPGAIN_DEFAULT // Estimation error parameter
-                                                // (Should be the same as Default) (1 in Q8)
-#define SUPGAIN_EPC_DT  200             // = SUPGAIN_ERROR_PARAM_C * ENERGY_DEV_TOL
-// Defines for "check delay estimation"
-#define CORR_WIDTH      31              // Number of samples to correlate over.
-#define CORR_MAX        16              // Maximum correlation offset
-#define CORR_MAX_BUF    63
-#define CORR_DEV        4
-#define CORR_MAX_LEVEL  20
-#define CORR_MAX_LOW    4
-#define CORR_BUF_LEN    (CORR_MAX << 1) + 1
-// Note that CORR_WIDTH + 2*CORR_MAX <= MAX_BUF_LEN
-
-#define ONE_Q14         (1 << 14)
-
-// NLP defines
-#define NLP_COMP_LOW    3277            // 0.2 in Q14
-#define NLP_COMP_HIGH   ONE_Q14         // 1 in Q14
-
-extern const WebRtc_Word16 WebRtcAecm_kSqrtHanning[];
-
-typedef struct {
-    WebRtc_Word16 real;
-    WebRtc_Word16 imag;
-} complex16_t;
-
-typedef struct
-{
-    int farBufWritePos;
-    int farBufReadPos;
-    int knownDelay;
-    int lastKnownDelay;
-    int firstVAD; // Parameter to control poorly initialized channels
-
-    void *farFrameBuf;
-    void *nearNoisyFrameBuf;
-    void *nearCleanFrameBuf;
-    void *outFrameBuf;
-
-    WebRtc_Word16 farBuf[FAR_BUF_LEN];
-
-    WebRtc_Word16 mult;
-    WebRtc_UWord32 seed;
-
-    // Delay estimation variables
-    void* delay_estimator;
-    WebRtc_UWord16 currentDelay;
-
-    WebRtc_Word16 nlpFlag;
-    WebRtc_Word16 fixedDelay;
-
-    WebRtc_UWord32 totCount;
-
-    WebRtc_Word16 dfaCleanQDomain;
-    WebRtc_Word16 dfaCleanQDomainOld;
-    WebRtc_Word16 dfaNoisyQDomain;
-    WebRtc_Word16 dfaNoisyQDomainOld;
-
-    WebRtc_Word16 nearLogEnergy[MAX_BUF_LEN];
-    WebRtc_Word16 farLogEnergy;
-    WebRtc_Word16 echoAdaptLogEnergy[MAX_BUF_LEN];
-    WebRtc_Word16 echoStoredLogEnergy[MAX_BUF_LEN];
-
-    // The extra 16 or 32 bytes in the following buffers are for alignment based Neon code.
-    // It's designed this way since the current GCC compiler can't align a buffer in 16 or 32
-    // byte boundaries properly.
-    WebRtc_Word16 channelStored_buf[PART_LEN1 + 8];
-    WebRtc_Word16 channelAdapt16_buf[PART_LEN1 + 8];
-    WebRtc_Word32 channelAdapt32_buf[PART_LEN1 + 8];
-    WebRtc_Word16 xBuf_buf[PART_LEN2 + 16]; // farend
-    WebRtc_Word16 dBufClean_buf[PART_LEN2 + 16]; // nearend
-    WebRtc_Word16 dBufNoisy_buf[PART_LEN2 + 16]; // nearend
-    WebRtc_Word16 outBuf_buf[PART_LEN + 8];
-
-    // Pointers to the above buffers
-    WebRtc_Word16 *channelStored;
-    WebRtc_Word16 *channelAdapt16;
-    WebRtc_Word32 *channelAdapt32;
-    WebRtc_Word16 *xBuf;
-    WebRtc_Word16 *dBufClean;
-    WebRtc_Word16 *dBufNoisy;
-    WebRtc_Word16 *outBuf;
-
-    WebRtc_Word32 echoFilt[PART_LEN1];
-    WebRtc_Word16 nearFilt[PART_LEN1];
-    WebRtc_Word32 noiseEst[PART_LEN1];
-    int           noiseEstTooLowCtr[PART_LEN1];
-    int           noiseEstTooHighCtr[PART_LEN1];
-    WebRtc_Word16 noiseEstCtr;
-    WebRtc_Word16 cngMode;
-
-    WebRtc_Word32 mseAdaptOld;
-    WebRtc_Word32 mseStoredOld;
-    WebRtc_Word32 mseThreshold;
-
-    WebRtc_Word16 farEnergyMin;
-    WebRtc_Word16 farEnergyMax;
-    WebRtc_Word16 farEnergyMaxMin;
-    WebRtc_Word16 farEnergyVAD;
-    WebRtc_Word16 farEnergyMSE;
-    int currentVADValue;
-    WebRtc_Word16 vadUpdateCount;
-
-    WebRtc_Word16 startupState;
-    WebRtc_Word16 mseChannelCount;
-    WebRtc_Word16 supGain;
-    WebRtc_Word16 supGainOld;
-
-    WebRtc_Word16 supGainErrParamA;
-    WebRtc_Word16 supGainErrParamD;
-    WebRtc_Word16 supGainErrParamDiffAB;
-    WebRtc_Word16 supGainErrParamDiffBD;
-
-#ifdef AEC_DEBUG
-    FILE *farFile;
-    FILE *nearFile;
-    FILE *outFile;
-#endif
-} AecmCore_t;
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_CreateCore(...)
-//
-// Allocates the memory needed by the AECM. The memory needs to be
-// initialized separately using the WebRtcAecm_InitCore() function.
-//
-// Input:
-//      - aecm          : Instance that should be created
-//
-// Output:
-//      - aecm          : Created instance
-//
-// Return value         :  0 - Ok
-//                        -1 - Error
-//
-int WebRtcAecm_CreateCore(AecmCore_t **aecm);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_InitCore(...)
-//
-// This function initializes the AECM instant created with WebRtcAecm_CreateCore(...)
-// Input:
-//      - aecm          : Pointer to the AECM instance
-//      - samplingFreq  : Sampling Frequency
-//
-// Output:
-//      - aecm          : Initialized instance
-//
-// Return value         :  0 - Ok
-//                        -1 - Error
-//
-int WebRtcAecm_InitCore(AecmCore_t * const aecm, int samplingFreq);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_FreeCore(...)
-//
-// This function releases the memory allocated by WebRtcAecm_CreateCore()
-// Input:
-//      - aecm          : Pointer to the AECM instance
-//
-// Return value         :  0 - Ok
-//                        -1 - Error
-//           11001-11016: Error
-//
-int WebRtcAecm_FreeCore(AecmCore_t *aecm);
-
-int WebRtcAecm_Control(AecmCore_t *aecm, int delay, int nlpFlag);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_InitEchoPathCore(...)
-//
-// This function resets the echo channel adaptation with the specified channel.
-// Input:
-//      - aecm          : Pointer to the AECM instance
-//      - echo_path     : Pointer to the data that should initialize the echo path
-//
-// Output:
-//      - aecm          : Initialized instance
-//
-void WebRtcAecm_InitEchoPathCore(AecmCore_t* aecm, const WebRtc_Word16* echo_path);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_ProcessFrame(...)
-//
-// This function processes frames and sends blocks to WebRtcAecm_ProcessBlock(...)
-//
-// Inputs:
-//      - aecm          : Pointer to the AECM instance
-//      - farend        : In buffer containing one frame of echo signal
-//      - nearendNoisy  : In buffer containing one frame of nearend+echo signal without NS
-//      - nearendClean  : In buffer containing one frame of nearend+echo signal with NS
-//
-// Output:
-//      - out           : Out buffer, one frame of nearend signal          :
-//
-//
-int WebRtcAecm_ProcessFrame(AecmCore_t * aecm, const WebRtc_Word16 * farend,
-                            const WebRtc_Word16 * nearendNoisy,
-                            const WebRtc_Word16 * nearendClean,
-                            WebRtc_Word16 * out);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_ProcessBlock(...)
-//
-// This function is called for every block within one frame
-// This function is called by WebRtcAecm_ProcessFrame(...)
-//
-// Inputs:
-//      - aecm          : Pointer to the AECM instance
-//      - farend        : In buffer containing one block of echo signal
-//      - nearendNoisy  : In buffer containing one frame of nearend+echo signal without NS
-//      - nearendClean  : In buffer containing one frame of nearend+echo signal with NS
-//
-// Output:
-//      - out           : Out buffer, one block of nearend signal          :
-//
-//
-int WebRtcAecm_ProcessBlock(AecmCore_t * aecm, const WebRtc_Word16 * farend,
-                            const WebRtc_Word16 * nearendNoisy,
-                            const WebRtc_Word16 * noisyClean,
-                            WebRtc_Word16 * out);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_BufferFarFrame()
-//
-// Inserts a frame of data into farend buffer.
-//
-// Inputs:
-//      - aecm          : Pointer to the AECM instance
-//      - farend        : In buffer containing one frame of farend signal
-//      - farLen        : Length of frame
-//
-void WebRtcAecm_BufferFarFrame(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
-                               const int farLen);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_FetchFarFrame()
-//
-// Read the farend buffer to account for known delay
-//
-// Inputs:
-//      - aecm          : Pointer to the AECM instance
-//      - farend        : In buffer containing one frame of farend signal
-//      - farLen        : Length of frame
-//      - knownDelay    : known delay
-//
-void WebRtcAecm_FetchFarFrame(AecmCore_t * const aecm, WebRtc_Word16 * const farend,
-                              const int farLen, const int knownDelay);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// Some internal functions shared by ARM NEON and generic C code:
-//
-
-void WebRtcAecm_CalcLinearEnergies(AecmCore_t* aecm,
-                                   const WebRtc_UWord16* far_spectrum,
-                                   WebRtc_Word32* echoEst,
-                                   WebRtc_UWord32* far_energy,
-                                   WebRtc_UWord32* echo_energy_adapt,
-                                   WebRtc_UWord32* echo_energy_stored);
-
-void WebRtcAecm_StoreAdaptiveChannel(AecmCore_t* aecm,
-                                     const WebRtc_UWord16* far_spectrum,
-                                     WebRtc_Word32* echo_est);
-
-void WebRtcAecm_ResetAdaptiveChannel(AecmCore_t *aecm);
-
-void WebRtcAecm_WindowAndFFT(WebRtc_Word16* fft,
-                             const WebRtc_Word16* time_signal,
-                             complex16_t* freq_signal,
-                             int time_signal_scaling);
-
-void WebRtcAecm_InverseFFTAndWindow(AecmCore_t* aecm,
-                                    WebRtc_Word16* fft,
-                                    complex16_t* efw,
-                                    WebRtc_Word16* output,
-                                    const WebRtc_Word16* nearendClean);
-
-#endif

src/modules/audio_processing/aecm/aecm_core_neon.c

@@ -1,314 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON)
-
-#include "aecm_core.h"
-
-#include <arm_neon.h>
-#include <assert.h>
-
-
-// Square root of Hanning window in Q14.
-static const WebRtc_Word16 kSqrtHanningReversed[] __attribute__ ((aligned (8))) = {       
-     16384, 16373, 16354, 16325,  
-     16286, 16237, 16179, 16111,  
-     16034, 15947, 15851, 15746,  
-     15631, 15506, 15373, 15231,  
-     15079, 14918, 14749, 14571,  
-     14384, 14189, 13985, 13773,  
-     13553, 13325, 13089, 12845,  
-     12594, 12335, 12068, 11795,  
-     11514, 11227, 10933, 10633,  
-     10326, 10013, 9695,  9370,   
-     9040,  8705,  8364,  8019,   
-     7668,  7313,  6954,  6591,   
-     6224,  5853,  5478,  5101,   
-     4720,  4337,  3951,  3562,   
-     3172,  2780,  2386,  1990,   
-     1594,  1196,  798,   399
-};
-
-void WebRtcAecm_WindowAndFFT(WebRtc_Word16* fft,
-                             const WebRtc_Word16* time_signal,
-                             complex16_t* freq_signal,
-                             int time_signal_scaling)
-{
-    int i, j;
-
-    int16x4_t tmp16x4_scaling = vdup_n_s16(time_signal_scaling);
-    __asm__("vmov.i16 d21, #0" ::: "d21");
-
-    for(i = 0, j = 0; i < PART_LEN; i += 4, j += 8)
-    {
-        int16x4_t tmp16x4_0;
-        int16x4_t tmp16x4_1;
-        int32x4_t tmp32x4_0;
-
-        /* Window near end */
-        // fft[j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((time_signal[i]
-        //       << time_signal_scaling), WebRtcAecm_kSqrtHanning[i], 14);
-        __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&time_signal[i]));
-        tmp16x4_0 = vshl_s16(tmp16x4_0, tmp16x4_scaling);
-
-        __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&WebRtcAecm_kSqrtHanning[i]));
-        tmp32x4_0 = vmull_s16(tmp16x4_0, tmp16x4_1);
-
-        __asm__("vshrn.i32 d20, %q0, #14" : : "w"(tmp32x4_0) : "d20");
-        __asm__("vst2.16 {d20, d21}, [%0, :128]" : : "r"(&fft[j]) : "q10");
-
-        // fft[PART_LEN2 + j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
-        //      (time_signal[PART_LEN + i] << time_signal_scaling),
-        //       WebRtcAecm_kSqrtHanning[PART_LEN - i], 14);
-        __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&time_signal[i + PART_LEN]));
-        tmp16x4_0 = vshl_s16(tmp16x4_0, tmp16x4_scaling);
-
-        __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&kSqrtHanningReversed[i]));
-        tmp32x4_0 = vmull_s16(tmp16x4_0, tmp16x4_1);
-
-        __asm__("vshrn.i32 d20, %q0, #14" : : "w"(tmp32x4_0) : "d20");
-        __asm__("vst2.16 {d20, d21}, [%0, :128]" : : "r"(&fft[PART_LEN2 + j]) : "q10");
-    }
-
-    WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
-    WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
-
-    // Take only the first PART_LEN2 samples, and switch the sign of the imaginary part.
-    for(i = 0, j = 0; j < PART_LEN2; i += 8, j += 16)
-    {
-        __asm__("vld2.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&fft[j]) : "q10", "q11");
-        __asm__("vneg.s16 d22, d22" : : : "q10");
-        __asm__("vneg.s16 d23, d23" : : : "q11");
-        __asm__("vst2.16 {d20, d21, d22, d23}, [%0, :256]" : : 
-            "r"(&freq_signal[i].real): "q10", "q11");
-    }
-}
-
-void WebRtcAecm_InverseFFTAndWindow(AecmCore_t* aecm,
-                        WebRtc_Word16* fft,
-                        complex16_t* efw,
-                        WebRtc_Word16* output,
-                        const WebRtc_Word16* nearendClean)
-{
-    int i, j, outCFFT;
-    WebRtc_Word32 tmp32no1;
-
-    // Synthesis
-    for(i = 0, j = 0; i < PART_LEN; i += 4, j += 8)
-    {
-        // We overwrite two more elements in fft[], but it's ok.
-        __asm__("vld2.16 {d20, d21}, [%0, :128]" : : "r"(&(efw[i].real)) : "q10");
-        __asm__("vmov q11, q10" : : : "q10", "q11");
-
-        __asm__("vneg.s16 d23, d23" : : : "q11");
-        __asm__("vst2.16 {d22, d23}, [%0, :128]" : : "r"(&fft[j]): "q11");
-
-        __asm__("vrev64.16 q10, q10" : : : "q10");
-        __asm__("vst2.16 {d20, d21}, [%0]" : : "r"(&fft[PART_LEN4 - j - 6]): "q10");
-    }
-
-    fft[PART_LEN2] = efw[PART_LEN].real;
-    fft[PART_LEN2 + 1] = -efw[PART_LEN].imag;
-
-    // Inverse FFT, result should be scaled with outCFFT.
-    WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
-    outCFFT = WebRtcSpl_ComplexIFFT(fft, PART_LEN_SHIFT, 1);
-
-    // Take only the real values and scale with outCFFT.
-    for (i = 0, j = 0; i < PART_LEN2; i += 8, j+= 16)
-    {
-        __asm__("vld2.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&fft[j]) : "q10", "q11");
-        __asm__("vst1.16 {d20, d21}, [%0, :128]" : : "r"(&fft[i]): "q10");
-    }
-
-    int32x4_t tmp32x4_2;
-    __asm__("vdup.32 %q0, %1" : "=w"(tmp32x4_2) : "r"((WebRtc_Word32)
-        (outCFFT - aecm->dfaCleanQDomain)));
-    for (i = 0; i < PART_LEN; i += 4)
-    {
-        int16x4_t tmp16x4_0;
-        int16x4_t tmp16x4_1;
-        int32x4_t tmp32x4_0;
-        int32x4_t tmp32x4_1;
-
-        // fft[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
-        //        fft[i], WebRtcAecm_kSqrtHanning[i], 14);
-        __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&fft[i]));
-        __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&WebRtcAecm_kSqrtHanning[i]));
-        __asm__("vmull.s16 %q0, %P1, %P2" : "=w"(tmp32x4_0) : "w"(tmp16x4_0), "w"(tmp16x4_1));
-        __asm__("vrshr.s32 %q0, %q1, #14" : "=w"(tmp32x4_0) : "0"(tmp32x4_0));
-
-        // tmp32no1 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)fft[i],
-        //        outCFFT - aecm->dfaCleanQDomain);
-        __asm__("vshl.s32 %q0, %q1, %q2" : "=w"(tmp32x4_0) : "0"(tmp32x4_0), "w"(tmp32x4_2));
-
-        // fft[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
-        //        tmp32no1 + outBuf[i], WEBRTC_SPL_WORD16_MIN);
-        // output[i] = fft[i];
-        __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&aecm->outBuf[i]));
-        __asm__("vmovl.s16 %q0, %P1" : "=w"(tmp32x4_1) : "w"(tmp16x4_0));
-        __asm__("vadd.i32 %q0, %q1" : : "w"(tmp32x4_0), "w"(tmp32x4_1));
-        __asm__("vqshrn.s32 %P0, %q1, #0" : "=w"(tmp16x4_0) : "w"(tmp32x4_0));
-        __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&fft[i]));
-        __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&output[i]));
-
-        // tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT(
-        //        fft[PART_LEN + i], WebRtcAecm_kSqrtHanning[PART_LEN - i], 14);
-        __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&fft[PART_LEN + i]));
-        __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&kSqrtHanningReversed[i]));
-        __asm__("vmull.s16 %q0, %P1, %P2" : "=w"(tmp32x4_0) : "w"(tmp16x4_0), "w"(tmp16x4_1));
-        __asm__("vshr.s32 %q0, %q1, #14" : "=w"(tmp32x4_0) : "0"(tmp32x4_0));
-
-        // tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, outCFFT - aecm->dfaCleanQDomain);
-        __asm__("vshl.s32 %q0, %q1, %q2" : "=w"(tmp32x4_0) : "0"(tmp32x4_0), "w"(tmp32x4_2));
-        // outBuf[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(
-        //        WEBRTC_SPL_WORD16_MAX, tmp32no1, WEBRTC_SPL_WORD16_MIN);
-        __asm__("vqshrn.s32 %P0, %q1, #0" : "=w"(tmp16x4_0) : "w"(tmp32x4_0));
-        __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&aecm->outBuf[i]));
-    }
-
-    // Copy the current block to the old position (outBuf is shifted elsewhere).
-    for (i = 0; i < PART_LEN; i += 16)
-    {
-        __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
-            "r"(&aecm->xBuf[i + PART_LEN]) : "q10");
-        __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&aecm->xBuf[i]): "q10");
-    }
-    for (i = 0; i < PART_LEN; i += 16)
-    {
-        __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
-            "r"(&aecm->dBufNoisy[i + PART_LEN]) : "q10");
-        __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : : 
-            "r"(&aecm->dBufNoisy[i]): "q10");
-    }
-    if (nearendClean != NULL) {
-        for (i = 0; i < PART_LEN; i += 16)
-        {
-            __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
-                "r"(&aecm->dBufClean[i + PART_LEN]) : "q10");
-            __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
-                "r"(&aecm->dBufClean[i]): "q10");
-        }
-    }
-}
-
-void WebRtcAecm_CalcLinearEnergies(AecmCore_t* aecm,
-                                   const WebRtc_UWord16* far_spectrum,
-                                   WebRtc_Word32* echo_est,
-                                   WebRtc_UWord32* far_energy,
-                                   WebRtc_UWord32* echo_energy_adapt,
-                                   WebRtc_UWord32* echo_energy_stored)
-{
-    int i;
-
-    register WebRtc_UWord32 far_energy_r;
-    register WebRtc_UWord32 echo_energy_stored_r;
-    register WebRtc_UWord32 echo_energy_adapt_r;
-    uint32x4_t tmp32x4_0;
-
-    __asm__("vmov.i32 q14, #0" : : : "q14"); // far_energy
-    __asm__("vmov.i32 q8,  #0" : : : "q8"); // echo_energy_stored
-    __asm__("vmov.i32 q9,  #0" : : : "q9"); // echo_energy_adapt
-
-    for(i = 0; i < PART_LEN -7; i += 8)
-    {
-        // far_energy += (WebRtc_UWord32)(far_spectrum[i]);
-        __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&far_spectrum[i]) : "q13");
-        __asm__("vaddw.u16 q14, q14, d26" : : : "q14", "q13");
-        __asm__("vaddw.u16 q14, q14, d27" : : : "q14", "q13");
-
-        // Get estimated echo energies for adaptive channel and stored channel.
-        // echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
-        __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelStored[i]) : "q12");
-        __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
-        __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
-        __asm__("vst1.32 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&echo_est[i]):
-            "q10", "q11");
-
-        // echo_energy_stored += (WebRtc_UWord32)echoEst[i];
-        __asm__("vadd.u32 q8, q10" : : : "q10", "q8");
-        __asm__("vadd.u32 q8, q11" : : : "q11", "q8");
-
-        // echo_energy_adapt += WEBRTC_SPL_UMUL_16_16(
-        //     aecm->channelAdapt16[i], far_spectrum[i]);
-        __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelAdapt16[i]) : "q12");
-        __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
-        __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
-        __asm__("vadd.u32 q9, q10" : : : "q9", "q15");
-        __asm__("vadd.u32 q9, q11" : : : "q9", "q11");
-    }
-
-    __asm__("vadd.u32 d28, d29" : : : "q14");
-    __asm__("vpadd.u32 d28, d28" : : : "q14");
-    __asm__("vmov.32 %0, d28[0]" : "=r"(far_energy_r): : "q14");
-
-    __asm__("vadd.u32 d18, d19" : : : "q9");
-    __asm__("vpadd.u32 d18, d18" : : : "q9");
-    __asm__("vmov.32 %0, d18[0]" : "=r"(echo_energy_adapt_r): : "q9");
-
-    __asm__("vadd.u32 d16, d17" : : : "q8");
-    __asm__("vpadd.u32 d16, d16" : : : "q8");
-    __asm__("vmov.32 %0, d16[0]" : "=r"(echo_energy_stored_r): : "q8");
-
-    // Get estimated echo energies for adaptive channel and stored channel.
-    echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
-    *echo_energy_stored = echo_energy_stored_r + (WebRtc_UWord32)echo_est[i];
-    *far_energy = far_energy_r + (WebRtc_UWord32)(far_spectrum[i]);
-    *echo_energy_adapt = echo_energy_adapt_r + WEBRTC_SPL_UMUL_16_16(
-        aecm->channelAdapt16[i], far_spectrum[i]);
-}
-
-void WebRtcAecm_StoreAdaptiveChannel(AecmCore_t* aecm,
-                                     const WebRtc_UWord16* far_spectrum,
-                                     WebRtc_Word32* echo_est)
-{
-    int i;
-
-    // During startup we store the channel every block.
-    // Recalculate echo estimate.
-    for(i = 0; i < PART_LEN -7; i += 8)
-    {
-        // aecm->channelStored[i] = acem->channelAdapt16[i];
-        // echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
-        __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&far_spectrum[i]) : "q13");
-        __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelAdapt16[i]) : "q12");
-        __asm__("vst1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelStored[i]) : "q12");
-        __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
-        __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
-        __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
-                               "r"(&echo_est[i]) : "q10", "q11");
-    }
-    aecm->channelStored[i] = aecm->channelAdapt16[i];
-    echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
-}
-
-void WebRtcAecm_ResetAdaptiveChannel(AecmCore_t* aecm)
-{
-    int i;
-
-    for(i = 0; i < PART_LEN -7; i += 8)
-    {
-        // aecm->channelAdapt16[i] = aecm->channelStored[i];
-        // aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)
-        //                           aecm->channelStored[i], 16);
-        __asm__("vld1.16 {d24, d25}, [%0, :128]" : :
-                        "r"(&aecm->channelStored[i]) : "q12");
-        __asm__("vst1.16 {d24, d25}, [%0, :128]" : :
-                        "r"(&aecm->channelAdapt16[i]) : "q12");
-        __asm__("vshll.s16 q10, d24, #16" : : : "q12", "q13", "q10");
-        __asm__("vshll.s16 q11, d25, #16" : : : "q12", "q13", "q11");
-        __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
-                        "r"(&aecm->channelAdapt32[i]): "q10", "q11");
-    }
-    aecm->channelAdapt16[i] = aecm->channelStored[i];
-    aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
-            (WebRtc_Word32)aecm->channelStored[i], 16);
-}
-
-#endif // #if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON)

src/modules/audio_processing/aecm/echo_control_mobile.c

@@ -1,800 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-//#include <string.h>
-
-#include "echo_control_mobile.h"
-#include "aecm_core.h"
-#include "ring_buffer.h"
-#ifdef AEC_DEBUG
-#include <stdio.h>
-#endif
-#ifdef MAC_IPHONE_PRINT
-#include <time.h>
-#include <stdio.h>
-#elif defined ARM_WINM_LOG
-#include "windows.h"
-extern HANDLE logFile;
-#endif
-
-#define BUF_SIZE_FRAMES 50 // buffer size (frames)
-// Maximum length of resampled signal. Must be an integer multiple of frames
-// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
-// The factor of 2 handles wb, and the + 1 is as a safety margin
-#define MAX_RESAMP_LEN (5 * FRAME_LEN)
-
-static const int kBufSizeSamp = BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
-static const int kSampMsNb = 8; // samples per ms in nb
-// Target suppression levels for nlp modes
-// log{0.001, 0.00001, 0.00000001}
-static const int kInitCheck = 42;
-
-typedef struct
-{
-    int sampFreq;
-    int scSampFreq;
-    short bufSizeStart;
-    int knownDelay;
-
-    // Stores the last frame added to the farend buffer
-    short farendOld[2][FRAME_LEN];
-    short initFlag; // indicates if AEC has been initialized
-
-    // Variables used for averaging far end buffer size
-    short counter;
-    short sum;
-    short firstVal;
-    short checkBufSizeCtr;
-
-    // Variables used for delay shifts
-    short msInSndCardBuf;
-    short filtDelay;
-    int timeForDelayChange;
-    int ECstartup;
-    int checkBuffSize;
-    int delayChange;
-    short lastDelayDiff;
-
-    WebRtc_Word16 echoMode;
-
-#ifdef AEC_DEBUG
-    FILE *bufFile;
-    FILE *delayFile;
-    FILE *preCompFile;
-    FILE *postCompFile;
-#endif // AEC_DEBUG
-    // Structures
-    void *farendBuf;
-
-    int lastError;
-
-    AecmCore_t *aecmCore;
-} aecmob_t;
-
-// Estimates delay to set the position of the farend buffer read pointer
-// (controlled by knownDelay)
-static int WebRtcAecm_EstBufDelay(aecmob_t *aecmInst, short msInSndCardBuf);
-
-// Stuffs the farend buffer if the estimated delay is too large
-static int WebRtcAecm_DelayComp(aecmob_t *aecmInst);
-
-WebRtc_Word32 WebRtcAecm_Create(void **aecmInst)
-{
-    aecmob_t *aecm;
-    if (aecmInst == NULL)
-    {
-        return -1;
-    }
-
-    aecm = malloc(sizeof(aecmob_t));
-    *aecmInst = aecm;
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (WebRtcAecm_CreateCore(&aecm->aecmCore) == -1)
-    {
-        WebRtcAecm_Free(aecm);
-        aecm = NULL;
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aecm->farendBuf, kBufSizeSamp) == -1)
-    {
-        WebRtcAecm_Free(aecm);
-        aecm = NULL;
-        return -1;
-    }
-
-    aecm->initFlag = 0;
-    aecm->lastError = 0;
-
-#ifdef AEC_DEBUG
-    aecm->aecmCore->farFile = fopen("aecFar.pcm","wb");
-    aecm->aecmCore->nearFile = fopen("aecNear.pcm","wb");
-    aecm->aecmCore->outFile = fopen("aecOut.pcm","wb");
-    //aecm->aecmCore->outLpFile = fopen("aecOutLp.pcm","wb");
-
-    aecm->bufFile = fopen("aecBuf.dat", "wb");
-    aecm->delayFile = fopen("aecDelay.dat", "wb");
-    aecm->preCompFile = fopen("preComp.pcm", "wb");
-    aecm->postCompFile = fopen("postComp.pcm", "wb");
-#endif // AEC_DEBUG
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_Free(void *aecmInst)
-{
-    aecmob_t *aecm = aecmInst;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-#ifdef AEC_DEBUG
-    fclose(aecm->aecmCore->farFile);
-    fclose(aecm->aecmCore->nearFile);
-    fclose(aecm->aecmCore->outFile);
-    //fclose(aecm->aecmCore->outLpFile);
-
-    fclose(aecm->bufFile);
-    fclose(aecm->delayFile);
-    fclose(aecm->preCompFile);
-    fclose(aecm->postCompFile);
-#endif // AEC_DEBUG
-    WebRtcAecm_FreeCore(aecm->aecmCore);
-    WebRtcApm_FreeBuffer(aecm->farendBuf);
-    free(aecm);
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_Init(void *aecmInst, WebRtc_Word32 sampFreq)
-{
-    aecmob_t *aecm = aecmInst;
-    AecmConfig aecConfig;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (sampFreq != 8000 && sampFreq != 16000)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecm->sampFreq = sampFreq;
-
-    // Initialize AECM core
-    if (WebRtcAecm_InitCore(aecm->aecmCore, aecm->sampFreq) == -1)
-    {
-        aecm->lastError = AECM_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    // Initialize farend buffer
-    if (WebRtcApm_InitBuffer(aecm->farendBuf) == -1)
-    {
-        aecm->lastError = AECM_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    aecm->initFlag = kInitCheck; // indicates that initialization has been done
-
-    aecm->delayChange = 1;
-
-    aecm->sum = 0;
-    aecm->counter = 0;
-    aecm->checkBuffSize = 1;
-    aecm->firstVal = 0;
-
-    aecm->ECstartup = 1;
-    aecm->bufSizeStart = 0;
-    aecm->checkBufSizeCtr = 0;
-    aecm->filtDelay = 0;
-    aecm->timeForDelayChange = 0;
-    aecm->knownDelay = 0;
-    aecm->lastDelayDiff = 0;
-
-    memset(&aecm->farendOld[0][0], 0, 160);
-
-    // Default settings.
-    aecConfig.cngMode = AecmTrue;
-    aecConfig.echoMode = 3;
-
-    if (WebRtcAecm_set_config(aecm, aecConfig) == -1)
-    {
-        aecm->lastError = AECM_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_BufferFarend(void *aecmInst, const WebRtc_Word16 *farend,
-                                      WebRtc_Word16 nrOfSamples)
-{
-    aecmob_t *aecm = aecmInst;
-    WebRtc_Word32 retVal = 0;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (farend == NULL)
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    if (nrOfSamples != 80 && nrOfSamples != 160)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-
-    // TODO: Is this really a good idea?
-    if (!aecm->ECstartup)
-    {
-        WebRtcAecm_DelayComp(aecm);
-    }
-
-    WebRtcApm_WriteBuffer(aecm->farendBuf, farend, nrOfSamples);
-
-    return retVal;
-}
-
-WebRtc_Word32 WebRtcAecm_Process(void *aecmInst, const WebRtc_Word16 *nearendNoisy,
-                                 const WebRtc_Word16 *nearendClean, WebRtc_Word16 *out,
-                                 WebRtc_Word16 nrOfSamples, WebRtc_Word16 msInSndCardBuf)
-{
-    aecmob_t *aecm = aecmInst;
-    WebRtc_Word32 retVal = 0;
-    short i;
-    short farend[FRAME_LEN];
-    short nmbrOfFilledBuffers;
-    short nBlocks10ms;
-    short nFrames;
-#ifdef AEC_DEBUG
-    short msInAECBuf;
-#endif
-
-#ifdef ARM_WINM_LOG
-    __int64 freq, start, end, diff;
-    unsigned int milliseconds;
-    DWORD temp;
-#elif defined MAC_IPHONE_PRINT
-    //       double endtime = 0, starttime = 0;
-    struct timeval starttime;
-    struct timeval endtime;
-    static long int timeused = 0;
-    static int timecount = 0;
-#endif
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (nearendNoisy == NULL)
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (out == NULL)
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    if (nrOfSamples != 80 && nrOfSamples != 160)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-
-    if (msInSndCardBuf < 0)
-    {
-        msInSndCardBuf = 0;
-        aecm->lastError = AECM_BAD_PARAMETER_WARNING;
-        retVal = -1;
-    } else if (msInSndCardBuf > 500)
-    {
-        msInSndCardBuf = 500;
-        aecm->lastError = AECM_BAD_PARAMETER_WARNING;
-        retVal = -1;
-    }
-    msInSndCardBuf += 10;
-    aecm->msInSndCardBuf = msInSndCardBuf;
-
-    nFrames = nrOfSamples / FRAME_LEN;
-    nBlocks10ms = nFrames / aecm->aecmCore->mult;
-
-    if (aecm->ECstartup)
-    {
-        if (nearendClean == NULL)
-        {
-            memcpy(out, nearendNoisy, sizeof(short) * nrOfSamples);
-        } else
-        {
-            memcpy(out, nearendClean, sizeof(short) * nrOfSamples);
-        }
-
-        nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecm->farendBuf) / FRAME_LEN;
-        // The AECM is in the start up mode
-        // AECM is disabled until the soundcard buffer and farend buffers are OK
-
-        // Mechanism to ensure that the soundcard buffer is reasonably stable.
-        if (aecm->checkBuffSize)
-        {
-            aecm->checkBufSizeCtr++;
-            // Before we fill up the far end buffer we require the amount of data on the
-            // sound card to be stable (+/-8 ms) compared to the first value. This
-            // comparison is made during the following 4 consecutive frames. If it seems
-            // to be stable then we start to fill up the far end buffer.
-
-            if (aecm->counter == 0)
-            {
-                aecm->firstVal = aecm->msInSndCardBuf;
-                aecm->sum = 0;
-            }
-
-            if (abs(aecm->firstVal - aecm->msInSndCardBuf)
-                    < WEBRTC_SPL_MAX(0.2 * aecm->msInSndCardBuf, kSampMsNb))
-            {
-                aecm->sum += aecm->msInSndCardBuf;
-                aecm->counter++;
-            } else
-            {
-                aecm->counter = 0;
-            }
-
-            if (aecm->counter * nBlocks10ms >= 6)
-            {
-                // The farend buffer size is determined in blocks of 80 samples
-                // Use 75% of the average value of the soundcard buffer
-                aecm->bufSizeStart
-                        = WEBRTC_SPL_MIN((3 * aecm->sum
-                                        * aecm->aecmCore->mult) / (aecm->counter * 40), BUF_SIZE_FRAMES);
-                // buffersize has now been determined
-                aecm->checkBuffSize = 0;
-            }
-
-            if (aecm->checkBufSizeCtr * nBlocks10ms > 50)
-            {
-                // for really bad sound cards, don't disable echocanceller for more than 0.5 sec
-                aecm->bufSizeStart = WEBRTC_SPL_MIN((3 * aecm->msInSndCardBuf
-                                * aecm->aecmCore->mult) / 40, BUF_SIZE_FRAMES);
-                aecm->checkBuffSize = 0;
-            }
-        }
-
-        // if checkBuffSize changed in the if-statement above
-        if (!aecm->checkBuffSize)
-        {
-            // soundcard buffer is now reasonably stable
-            // When the far end buffer is filled with approximately the same amount of
-            // data as the amount on the sound card we end the start up phase and start
-            // to cancel echoes.
-
-            if (nmbrOfFilledBuffers == aecm->bufSizeStart)
-            {
-                aecm->ECstartup = 0; // Enable the AECM
-            } else if (nmbrOfFilledBuffers > aecm->bufSizeStart)
-            {
-                WebRtcApm_FlushBuffer(
-                                       aecm->farendBuf,
-                                       WebRtcApm_get_buffer_size(aecm->farendBuf)
-                                               - aecm->bufSizeStart * FRAME_LEN);
-                aecm->ECstartup = 0;
-            }
-        }
-
-    } else
-    {
-        // AECM is enabled
-
-        // Note only 1 block supported for nb and 2 blocks for wb
-        for (i = 0; i < nFrames; i++)
-        {
-            nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecm->farendBuf) / FRAME_LEN;
-
-            // Check that there is data in the far end buffer
-            if (nmbrOfFilledBuffers > 0)
-            {
-                // Get the next 80 samples from the farend buffer
-                WebRtcApm_ReadBuffer(aecm->farendBuf, farend, FRAME_LEN);
-
-                // Always store the last frame for use when we run out of data
-                memcpy(&(aecm->farendOld[i][0]), farend, FRAME_LEN * sizeof(short));
-            } else
-            {
-                // We have no data so we use the last played frame
-                memcpy(farend, &(aecm->farendOld[i][0]), FRAME_LEN * sizeof(short));
-            }
-
-            // Call buffer delay estimator when all data is extracted,
-            // i,e. i = 0 for NB and i = 1 for WB
-            if ((i == 0 && aecm->sampFreq == 8000) || (i == 1 && aecm->sampFreq == 16000))
-            {
-                WebRtcAecm_EstBufDelay(aecm, aecm->msInSndCardBuf);
-            }
-
-#ifdef ARM_WINM_LOG
-            // measure tick start
-            QueryPerformanceFrequency((LARGE_INTEGER*)&freq);
-            QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#elif defined MAC_IPHONE_PRINT
-            //            starttime = clock()/(double)CLOCKS_PER_SEC;
-            gettimeofday(&starttime, NULL);
-#endif
-            // Call the AECM
-            /*WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearend[FRAME_LEN * i],
-             &out[FRAME_LEN * i], aecm->knownDelay);*/
-            if (nearendClean == NULL)
-            {
-                if (WebRtcAecm_ProcessFrame(aecm->aecmCore,
-                                            farend,
-                                            &nearendNoisy[FRAME_LEN * i],
-                                            NULL,
-                                            &out[FRAME_LEN * i]) == -1)
-                {
-                    return -1;
-                }
-            } else
-            {
-                if (WebRtcAecm_ProcessFrame(aecm->aecmCore,
-                                            farend,
-                                            &nearendNoisy[FRAME_LEN * i],
-                                            &nearendClean[FRAME_LEN * i],
-                                            &out[FRAME_LEN * i]) == -1)
-                {
-                    return -1;
-                }
-            }
-
-#ifdef ARM_WINM_LOG
-
-            // measure tick end
-            QueryPerformanceCounter((LARGE_INTEGER*)&end);
-
-            if(end > start)
-            {
-                diff = ((end - start) * 1000) / (freq/1000);
-                milliseconds = (unsigned int)(diff & 0xffffffff);
-                WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-            }
-#elif defined MAC_IPHONE_PRINT
-            //            endtime = clock()/(double)CLOCKS_PER_SEC;
-            //            printf("%f\n", endtime - starttime);
-
-            gettimeofday(&endtime, NULL);
-
-            if( endtime.tv_usec > starttime.tv_usec)
-            {
-                timeused += endtime.tv_usec - starttime.tv_usec;
-            } else
-            {
-                timeused += endtime.tv_usec + 1000000 - starttime.tv_usec;
-            }
-
-            if(++timecount == 1000)
-            {
-                timecount = 0;
-                printf("AEC: %ld\n", timeused);
-                timeused = 0;
-            }
-#endif
-
-        }
-    }
-
-#ifdef AEC_DEBUG
-    msInAECBuf = WebRtcApm_get_buffer_size(aecm->farendBuf) / (kSampMsNb*aecm->aecmCore->mult);
-    fwrite(&msInAECBuf, 2, 1, aecm->bufFile);
-    fwrite(&(aecm->knownDelay), sizeof(aecm->knownDelay), 1, aecm->delayFile);
-#endif
-
-    return retVal;
-}
-
-WebRtc_Word32 WebRtcAecm_set_config(void *aecmInst, AecmConfig config)
-{
-    aecmob_t *aecm = aecmInst;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    if (config.cngMode != AecmFalse && config.cngMode != AecmTrue)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecm->aecmCore->cngMode = config.cngMode;
-
-    if (config.echoMode < 0 || config.echoMode > 4)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecm->echoMode = config.echoMode;
-
-    if (aecm->echoMode == 0)
-    {
-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 3;
-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 3;
-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 3;
-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 3;
-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 3)
-                - (SUPGAIN_ERROR_PARAM_B >> 3);
-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 3)
-                - (SUPGAIN_ERROR_PARAM_D >> 3);
-    } else if (aecm->echoMode == 1)
-    {
-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 2;
-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 2;
-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 2;
-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 2;
-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 2)
-                - (SUPGAIN_ERROR_PARAM_B >> 2);
-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 2)
-                - (SUPGAIN_ERROR_PARAM_D >> 2);
-    } else if (aecm->echoMode == 2)
-    {
-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 1;
-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 1;
-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 1;
-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 1;
-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 1)
-                - (SUPGAIN_ERROR_PARAM_B >> 1);
-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 1)
-                - (SUPGAIN_ERROR_PARAM_D >> 1);
-    } else if (aecm->echoMode == 3)
-    {
-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT;
-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT;
-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
-        aecm->aecmCore->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
-        aecm->aecmCore->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
-    } else if (aecm->echoMode == 4)
-    {
-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT << 1;
-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT << 1;
-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A << 1;
-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D << 1;
-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A << 1)
-                - (SUPGAIN_ERROR_PARAM_B << 1);
-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B << 1)
-                - (SUPGAIN_ERROR_PARAM_D << 1);
-    }
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_get_config(void *aecmInst, AecmConfig *config)
-{
-    aecmob_t *aecm = aecmInst;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (config == NULL)
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    config->cngMode = aecm->aecmCore->cngMode;
-    config->echoMode = aecm->echoMode;
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_InitEchoPath(void* aecmInst,
-                                      const void* echo_path,
-                                      size_t size_bytes)
-{
-    aecmob_t *aecm = aecmInst;
-    const WebRtc_Word16* echo_path_ptr = echo_path;
-
-    if ((aecm == NULL) || (echo_path == NULL))
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
-    if (size_bytes != WebRtcAecm_echo_path_size_bytes())
-    {
-        // Input channel size does not match the size of AECM
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    WebRtcAecm_InitEchoPathCore(aecm->aecmCore, echo_path_ptr);
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_GetEchoPath(void* aecmInst,
-                                     void* echo_path,
-                                     size_t size_bytes)
-{
-    aecmob_t *aecm = aecmInst;
-    WebRtc_Word16* echo_path_ptr = echo_path;
-
-    if ((aecm == NULL) || (echo_path == NULL))
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
-    if (size_bytes != WebRtcAecm_echo_path_size_bytes())
-    {
-        // Input channel size does not match the size of AECM
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    memcpy(echo_path_ptr, aecm->aecmCore->channelStored, size_bytes);
-    return 0;
-}
-
-size_t WebRtcAecm_echo_path_size_bytes()
-{
-    return (PART_LEN1 * sizeof(WebRtc_Word16));
-}
-
-WebRtc_Word32 WebRtcAecm_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len)
-{
-    const char version[] = "AECM 1.2.0";
-    const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
-
-    if (versionStr == NULL)
-    {
-        return -1;
-    }
-
-    if (versionLen > len)
-    {
-        return -1;
-    }
-
-    strncpy(versionStr, version, versionLen);
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_get_error_code(void *aecmInst)
-{
-    aecmob_t *aecm = aecmInst;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    return aecm->lastError;
-}
-
-static int WebRtcAecm_EstBufDelay(aecmob_t *aecm, short msInSndCardBuf)
-{
-    short delayNew, nSampFar, nSampSndCard;
-    short diff;
-
-    nSampFar = WebRtcApm_get_buffer_size(aecm->farendBuf);
-    nSampSndCard = msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
-
-    delayNew = nSampSndCard - nSampFar;
-
-    if (delayNew < FRAME_LEN)
-    {
-        WebRtcApm_FlushBuffer(aecm->farendBuf, FRAME_LEN);
-        delayNew += FRAME_LEN;
-    }
-
-    aecm->filtDelay = WEBRTC_SPL_MAX(0, (8 * aecm->filtDelay + 2 * delayNew) / 10);
-
-    diff = aecm->filtDelay - aecm->knownDelay;
-    if (diff > 224)
-    {
-        if (aecm->lastDelayDiff < 96)
-        {
-            aecm->timeForDelayChange = 0;
-        } else
-        {
-            aecm->timeForDelayChange++;
-        }
-    } else if (diff < 96 && aecm->knownDelay > 0)
-    {
-        if (aecm->lastDelayDiff > 224)
-        {
-            aecm->timeForDelayChange = 0;
-        } else
-        {
-            aecm->timeForDelayChange++;
-        }
-    } else
-    {
-        aecm->timeForDelayChange = 0;
-    }
-    aecm->lastDelayDiff = diff;
-
-    if (aecm->timeForDelayChange > 25)
-    {
-        aecm->knownDelay = WEBRTC_SPL_MAX((int)aecm->filtDelay - 160, 0);
-    }
-    return 0;
-}
-
-static int WebRtcAecm_DelayComp(aecmob_t *aecm)
-{
-    int nSampFar, nSampSndCard, delayNew, nSampAdd;
-    const int maxStuffSamp = 10 * FRAME_LEN;
-
-    nSampFar = WebRtcApm_get_buffer_size(aecm->farendBuf);
-    nSampSndCard = aecm->msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
-    delayNew = nSampSndCard - nSampFar;
-
-    if (delayNew > FAR_BUF_LEN - FRAME_LEN * aecm->aecmCore->mult)
-    {
-        // The difference of the buffer sizes is larger than the maximum
-        // allowed known delay. Compensate by stuffing the buffer.
-        nSampAdd = (int)(WEBRTC_SPL_MAX(((nSampSndCard >> 1) - nSampFar),
-                FRAME_LEN));
-        nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
-
-        WebRtcApm_StuffBuffer(aecm->farendBuf, nSampAdd);
-        aecm->delayChange = 1; // the delay needs to be updated
-    }
-
-    return 0;
-}

src/modules/audio_processing/aecm/interface/echo_control_mobile.h

@@ -1,250 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_INTERFACE_ECHO_CONTROL_MOBILE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_INTERFACE_ECHO_CONTROL_MOBILE_H_
-
-#include "typedefs.h"
-
-enum {
-    AecmFalse = 0,
-    AecmTrue
-};
-
-// Errors
-#define AECM_UNSPECIFIED_ERROR           12000
-#define AECM_UNSUPPORTED_FUNCTION_ERROR  12001
-#define AECM_UNINITIALIZED_ERROR         12002
-#define AECM_NULL_POINTER_ERROR          12003
-#define AECM_BAD_PARAMETER_ERROR         12004
-
-// Warnings
-#define AECM_BAD_PARAMETER_WARNING       12100
-
-typedef struct {
-    WebRtc_Word16 cngMode;            // AECM_FALSE, AECM_TRUE (default)
-    WebRtc_Word16 echoMode;           // 0, 1, 2, 3 (default), 4
-} AecmConfig;
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * Allocates the memory needed by the AECM. The memory needs to be
- * initialized separately using the WebRtcAecm_Init() function.
- *
- * Inputs                           Description
- * -------------------------------------------------------------------
- * void **aecmInst                  Pointer to the AECM instance to be
- *                                  created and initialized
- *
- * Outputs                          Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return             0: OK
- *                                 -1: error
- */
-WebRtc_Word32 WebRtcAecm_Create(void **aecmInst);
-
-/*
- * This function releases the memory allocated by WebRtcAecm_Create()
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void *aecmInst               Pointer to the AECM instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_Free(void *aecmInst);
-
-/*
- * Initializes an AECM instance.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecmInst     Pointer to the AECM instance
- * WebRtc_Word32  sampFreq      Sampling frequency of data
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_Init(void* aecmInst,
-                              WebRtc_Word32 sampFreq);
-
-/*
- * Inserts an 80 or 160 sample block of data into the farend buffer.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecmInst     Pointer to the AECM instance
- * WebRtc_Word16  *farend       In buffer containing one frame of
- *                              farend signal
- * WebRtc_Word16  nrOfSamples   Number of samples in farend buffer
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_BufferFarend(void* aecmInst,
-                                      const WebRtc_Word16* farend,
-                                      WebRtc_Word16 nrOfSamples);
-
-/*
- * Runs the AECM on an 80 or 160 sample blocks of data.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecmInst      Pointer to the AECM instance
- * WebRtc_Word16  *nearendNoisy  In buffer containing one frame of
- *                               reference nearend+echo signal. If
- *                               noise reduction is active, provide
- *                               the noisy signal here.
- * WebRtc_Word16  *nearendClean  In buffer containing one frame of
- *                               nearend+echo signal. If noise
- *                               reduction is active, provide the
- *                               clean signal here. Otherwise pass a
- *                               NULL pointer.
- * WebRtc_Word16  nrOfSamples    Number of samples in nearend buffer
- * WebRtc_Word16  msInSndCardBuf Delay estimate for sound card and
- *                               system buffers
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word16  *out          Out buffer, one frame of processed nearend
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_Process(void* aecmInst,
-                                 const WebRtc_Word16* nearendNoisy,
-                                 const WebRtc_Word16* nearendClean,
-                                 WebRtc_Word16* out,
-                                 WebRtc_Word16 nrOfSamples,
-                                 WebRtc_Word16 msInSndCardBuf);
-
-/*
- * This function enables the user to set certain parameters on-the-fly
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void     *aecmInst           Pointer to the AECM instance
- * AecmConfig config            Config instance that contains all
- *                              properties to be set
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_set_config(void* aecmInst,
-                                    AecmConfig config);
-
-/*
- * This function enables the user to set certain parameters on-the-fly
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void *aecmInst               Pointer to the AECM instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * AecmConfig  *config          Pointer to the config instance that
- *                              all properties will be written to
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_get_config(void *aecmInst,
-                                    AecmConfig *config);
-
-/*
- * This function enables the user to set the echo path on-the-fly.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void*        aecmInst        Pointer to the AECM instance
- * void*        echo_path       Pointer to the echo path to be set
- * size_t       size_bytes      Size in bytes of the echo path
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_InitEchoPath(void* aecmInst,
-                                      const void* echo_path,
-                                      size_t size_bytes);
-
-/*
- * This function enables the user to get the currently used echo path
- * on-the-fly
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void*        aecmInst        Pointer to the AECM instance
- * void*        echo_path       Pointer to echo path
- * size_t       size_bytes      Size in bytes of the echo path
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_GetEchoPath(void* aecmInst,
-                                     void* echo_path,
-                                     size_t size_bytes);
-
-/*
- * This function enables the user to get the echo path size in bytes
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * size_t       return           : size in bytes
- */
-size_t WebRtcAecm_echo_path_size_bytes();
-
-/*
- * Gets the last error code.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void         *aecmInst       Pointer to the AECM instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        11000-11100: error code
- */
-WebRtc_Word32 WebRtcAecm_get_error_code(void *aecmInst);
-
-/*
- * Gets a version string
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * char           *versionStr   Pointer to a string array
- * WebRtc_Word16  len           The maximum length of the string
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word8   *versionStr   Pointer to a string array
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_get_version(WebRtc_Word8 *versionStr,
-                                     WebRtc_Word16 len);
-
-#ifdef __cplusplus
-}
-#endif
-#endif /* WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_INTERFACE_ECHO_CONTROL_MOBILE_H_ */

src/modules/audio_processing/agc/Makefile.am

@@ -1,10 +0,0 @@
-noinst_LTLIBRARIES = libagc.la
-
-libagc_la_SOURCES = interface/gain_control.h \
-		    analog_agc.c \
-		    analog_agc.h \
-		    digital_agc.c \
-		    digital_agc.h
-libagc_la_CFLAGS = $(AM_CFLAGS) $(COMMON_CFLAGS) \
-		   -I$(top_srcdir)/src/common_audio/signal_processing_library/main/interface \
-		   -I$(top_srcdir)/src/modules/audio_processing/utility

src/modules/audio_processing/agc/agc.gypi

@@ -1,34 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'targets': [
-    {
-      'target_name': 'agc',
-      'type': '<(library)',
-      'dependencies': [
-        '<(webrtc_root)/common_audio/common_audio.gyp:spl',
-      ],
-      'include_dirs': [
-        'interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'interface',
-        ],
-      },
-      'sources': [
-        'interface/gain_control.h',
-        'analog_agc.c',
-        'analog_agc.h',
-        'digital_agc.c',
-        'digital_agc.h',
-      ],
-    },
-  ],
-}

src/modules/audio_processing/agc/analog_agc.h

@@ -1,133 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
-
-#include "typedefs.h"
-#include "gain_control.h"
-#include "digital_agc.h"
-
-//#define AGC_DEBUG
-//#define MIC_LEVEL_FEEDBACK
-#ifdef AGC_DEBUG
-#include <stdio.h>
-#endif
-
-/* Analog Automatic Gain Control variables:
- * Constant declarations (inner limits inside which no changes are done)
- * In the beginning the range is narrower to widen as soon as the measure
- * 'Rxx160_LP' is inside it. Currently the starting limits are -22.2+/-1dBm0
- * and the final limits -22.2+/-2.5dBm0. These levels makes the speech signal
- * go towards -25.4dBm0 (-31.4dBov). Tuned with wbfile-31.4dBov.pcm
- * The limits are created by running the AGC with a file having the desired
- * signal level and thereafter plotting Rxx160_LP in the dBm0-domain defined
- * by out=10*log10(in/260537279.7); Set the target level to the average level
- * of our measure Rxx160_LP. Remember that the levels are in blocks of 16 in
- * Q(-7). (Example matlab code: round(db2pow(-21.2)*16/2^7) )
- */
-#define RXX_BUFFER_LEN  10
-
-static const WebRtc_Word16 kMsecSpeechInner = 520;
-static const WebRtc_Word16 kMsecSpeechOuter = 340;
-
-static const WebRtc_Word16 kNormalVadThreshold = 400;
-
-static const WebRtc_Word16 kAlphaShortTerm = 6; // 1 >> 6 = 0.0156
-static const WebRtc_Word16 kAlphaLongTerm = 10; // 1 >> 10 = 0.000977
-
-typedef struct
-{
-    // Configurable parameters/variables
-    WebRtc_UWord32      fs;                 // Sampling frequency
-    WebRtc_Word16       compressionGaindB;  // Fixed gain level in dB
-    WebRtc_Word16       targetLevelDbfs;    // Target level in -dBfs of envelope (default -3)
-    WebRtc_Word16       agcMode;            // Hard coded mode (adaptAna/adaptDig/fixedDig)
-    WebRtc_UWord8       limiterEnable;      // Enabling limiter (on/off (default off))
-    WebRtcAgc_config_t  defaultConfig;
-    WebRtcAgc_config_t  usedConfig;
-
-    // General variables
-    WebRtc_Word16       initFlag;
-    WebRtc_Word16       lastError;
-
-    // Target level parameters
-    // Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7)
-    WebRtc_Word32       analogTargetLevel;  // = RXX_BUFFER_LEN * 846805;       -22 dBfs
-    WebRtc_Word32       startUpperLimit;    // = RXX_BUFFER_LEN * 1066064;      -21 dBfs
-    WebRtc_Word32       startLowerLimit;    // = RXX_BUFFER_LEN * 672641;       -23 dBfs
-    WebRtc_Word32       upperPrimaryLimit;  // = RXX_BUFFER_LEN * 1342095;      -20 dBfs
-    WebRtc_Word32       lowerPrimaryLimit;  // = RXX_BUFFER_LEN * 534298;       -24 dBfs
-    WebRtc_Word32       upperSecondaryLimit;// = RXX_BUFFER_LEN * 2677832;      -17 dBfs
-    WebRtc_Word32       lowerSecondaryLimit;// = RXX_BUFFER_LEN * 267783;       -27 dBfs
-    WebRtc_UWord16      targetIdx;          // Table index for corresponding target level
-#ifdef MIC_LEVEL_FEEDBACK
-    WebRtc_UWord16      targetIdxOffset;    // Table index offset for level compensation
-#endif
-    WebRtc_Word16       analogTarget;       // Digital reference level in ENV scale
-
-    // Analog AGC specific variables
-    WebRtc_Word32       filterState[8];     // For downsampling wb to nb
-    WebRtc_Word32       upperLimit;         // Upper limit for mic energy
-    WebRtc_Word32       lowerLimit;         // Lower limit for mic energy
-    WebRtc_Word32       Rxx160w32;          // Average energy for one frame
-    WebRtc_Word32       Rxx16_LPw32;        // Low pass filtered subframe energies
-    WebRtc_Word32       Rxx160_LPw32;       // Low pass filtered frame energies
-    WebRtc_Word32       Rxx16_LPw32Max;     // Keeps track of largest energy subframe
-    WebRtc_Word32       Rxx16_vectorw32[RXX_BUFFER_LEN];// Array with subframe energies
-    WebRtc_Word32       Rxx16w32_array[2][5];// Energy values of microphone signal
-    WebRtc_Word32       env[2][10];         // Envelope values of subframes
-
-    WebRtc_Word16       Rxx16pos;           // Current position in the Rxx16_vectorw32
-    WebRtc_Word16       envSum;             // Filtered scaled envelope in subframes
-    WebRtc_Word16       vadThreshold;       // Threshold for VAD decision
-    WebRtc_Word16       inActive;           // Inactive time in milliseconds
-    WebRtc_Word16       msTooLow;           // Milliseconds of speech at a too low level
-    WebRtc_Word16       msTooHigh;          // Milliseconds of speech at a too high level
-    WebRtc_Word16       changeToSlowMode;   // Change to slow mode after some time at target
-    WebRtc_Word16       firstCall;          // First call to the process-function
-    WebRtc_Word16       msZero;             // Milliseconds of zero input
-    WebRtc_Word16       msecSpeechOuterChange;// Min ms of speech between volume changes
-    WebRtc_Word16       msecSpeechInnerChange;// Min ms of speech between volume changes
-    WebRtc_Word16       activeSpeech;       // Milliseconds of active speech
-    WebRtc_Word16       muteGuardMs;        // Counter to prevent mute action
-    WebRtc_Word16       inQueue;            // 10 ms batch indicator
-
-    // Microphone level variables
-    WebRtc_Word32       micRef;             // Remember ref. mic level for virtual mic
-    WebRtc_UWord16      gainTableIdx;       // Current position in virtual gain table
-    WebRtc_Word32       micGainIdx;         // Gain index of mic level to increase slowly
-    WebRtc_Word32       micVol;             // Remember volume between frames
-    WebRtc_Word32       maxLevel;           // Max possible vol level, incl dig gain
-    WebRtc_Word32       maxAnalog;          // Maximum possible analog volume level
-    WebRtc_Word32       maxInit;            // Initial value of "max"
-    WebRtc_Word32       minLevel;           // Minimum possible volume level
-    WebRtc_Word32       minOutput;          // Minimum output volume level
-    WebRtc_Word32       zeroCtrlMax;        // Remember max gain => don't amp low input
-
-    WebRtc_Word16       scale;              // Scale factor for internal volume levels
-#ifdef MIC_LEVEL_FEEDBACK
-    WebRtc_Word16       numBlocksMicLvlSat;
-    WebRtc_UWord8 micLvlSat;
-#endif
-    // Structs for VAD and digital_agc
-    AgcVad_t            vadMic;
-    DigitalAgc_t        digitalAgc;
-
-#ifdef AGC_DEBUG
-    FILE*               fpt;
-    FILE*               agcLog;
-    WebRtc_Word32       fcount;
-#endif
-
-    WebRtc_Word16       lowLevelSignal;
-} Agc_t;
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_

src/modules/audio_processing/agc/digital_agc.c

@@ -1,786 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/* digital_agc.c
- *
- */
-
-#include <string.h>
-#ifdef AGC_DEBUG
-#include <stdio.h>
-#endif
-#include "digital_agc.h"
-#include "gain_control.h"
-
-// To generate the gaintable, copy&paste the following lines to a Matlab window:
-// MaxGain = 6; MinGain = 0; CompRatio = 3; Knee = 1;
-// zeros = 0:31; lvl = 2.^(1-zeros);
-// A = -10*log10(lvl) * (CompRatio - 1) / CompRatio;
-// B = MaxGain - MinGain;
-// gains = round(2^16*10.^(0.05 * (MinGain + B * ( log(exp(-Knee*A)+exp(-Knee*B)) - log(1+exp(-Knee*B)) ) / log(1/(1+exp(Knee*B))))));
-// fprintf(1, '\t%i, %i, %i, %i,\n', gains);
-// % Matlab code for plotting the gain and input/output level characteristic (copy/paste the following 3 lines):
-// in = 10*log10(lvl); out = 20*log10(gains/65536);
-// subplot(121); plot(in, out); axis([-30, 0, -5, 20]); grid on; xlabel('Input (dB)'); ylabel('Gain (dB)');
-// subplot(122); plot(in, in+out); axis([-30, 0, -30, 5]); grid on; xlabel('Input (dB)'); ylabel('Output (dB)');
-// zoom on;
-
-// Generator table for y=log2(1+e^x) in Q8.
-static const WebRtc_UWord16 kGenFuncTable[128] = {
-          256,   485,   786,  1126,  1484,  1849,  2217,  2586,
-         2955,  3324,  3693,  4063,  4432,  4801,  5171,  5540,
-         5909,  6279,  6648,  7017,  7387,  7756,  8125,  8495,
-         8864,  9233,  9603,  9972, 10341, 10711, 11080, 11449,
-        11819, 12188, 12557, 12927, 13296, 13665, 14035, 14404,
-        14773, 15143, 15512, 15881, 16251, 16620, 16989, 17359,
-        17728, 18097, 18466, 18836, 19205, 19574, 19944, 20313,
-        20682, 21052, 21421, 21790, 22160, 22529, 22898, 23268,
-        23637, 24006, 24376, 24745, 25114, 25484, 25853, 26222,
-        26592, 26961, 27330, 27700, 28069, 28438, 28808, 29177,
-        29546, 29916, 30285, 30654, 31024, 31393, 31762, 32132,
-        32501, 32870, 33240, 33609, 33978, 34348, 34717, 35086,
-        35456, 35825, 36194, 36564, 36933, 37302, 37672, 38041,
-        38410, 38780, 39149, 39518, 39888, 40257, 40626, 40996,
-        41365, 41734, 42104, 42473, 42842, 43212, 43581, 43950,
-        44320, 44689, 45058, 45428, 45797, 46166, 46536, 46905
-};
-
-static const WebRtc_Word16 kAvgDecayTime = 250; // frames; < 3000
-
-WebRtc_Word32 WebRtcAgc_CalculateGainTable(WebRtc_Word32 *gainTable, // Q16
-                                           WebRtc_Word16 digCompGaindB, // Q0
-                                           WebRtc_Word16 targetLevelDbfs,// Q0
-                                           WebRtc_UWord8 limiterEnable,
-                                           WebRtc_Word16 analogTarget) // Q0
-{
-    // This function generates the compressor gain table used in the fixed digital part.
-    WebRtc_UWord32 tmpU32no1, tmpU32no2, absInLevel, logApprox;
-    WebRtc_Word32 inLevel, limiterLvl;
-    WebRtc_Word32 tmp32, tmp32no1, tmp32no2, numFIX, den, y32;
-    const WebRtc_UWord16 kLog10 = 54426; // log2(10)     in Q14
-    const WebRtc_UWord16 kLog10_2 = 49321; // 10*log10(2)  in Q14
-    const WebRtc_UWord16 kLogE_1 = 23637; // log2(e)      in Q14
-    WebRtc_UWord16 constMaxGain;
-    WebRtc_UWord16 tmpU16, intPart, fracPart;
-    const WebRtc_Word16 kCompRatio = 3;
-    const WebRtc_Word16 kSoftLimiterLeft = 1;
-    WebRtc_Word16 limiterOffset = 0; // Limiter offset
-    WebRtc_Word16 limiterIdx, limiterLvlX;
-    WebRtc_Word16 constLinApprox, zeroGainLvl, maxGain, diffGain;
-    WebRtc_Word16 i, tmp16, tmp16no1;
-    int zeros, zerosScale;
-
-    // Constants
-//    kLogE_1 = 23637; // log2(e)      in Q14
-//    kLog10 = 54426; // log2(10)     in Q14
-//    kLog10_2 = 49321; // 10*log10(2)  in Q14
-
-    // Calculate maximum digital gain and zero gain level
-    tmp32no1 = WEBRTC_SPL_MUL_16_16(digCompGaindB - analogTarget, kCompRatio - 1);
-    tmp16no1 = analogTarget - targetLevelDbfs;
-    tmp16no1 += WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
-    maxGain = WEBRTC_SPL_MAX(tmp16no1, (analogTarget - targetLevelDbfs));
-    tmp32no1 = WEBRTC_SPL_MUL_16_16(maxGain, kCompRatio);
-    zeroGainLvl = digCompGaindB;
-    zeroGainLvl -= WebRtcSpl_DivW32W16ResW16(tmp32no1 + ((kCompRatio - 1) >> 1),
-                                             kCompRatio - 1);
-    if ((digCompGaindB <= analogTarget) && (limiterEnable))
-    {
-        zeroGainLvl += (analogTarget - digCompGaindB + kSoftLimiterLeft);
-        limiterOffset = 0;
-    }
-
-    // Calculate the difference between maximum gain and gain at 0dB0v:
-    //  diffGain = maxGain + (compRatio-1)*zeroGainLvl/compRatio
-    //           = (compRatio-1)*digCompGaindB/compRatio
-    tmp32no1 = WEBRTC_SPL_MUL_16_16(digCompGaindB, kCompRatio - 1);
-    diffGain = WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
-    if (diffGain < 0)
-    {
-        return -1;
-    }
-
-    // Calculate the limiter level and index:
-    //  limiterLvlX = analogTarget - limiterOffset
-    //  limiterLvl  = targetLevelDbfs + limiterOffset/compRatio
-    limiterLvlX = analogTarget - limiterOffset;
-    limiterIdx = 2
-            + WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)limiterLvlX, 13),
-                                        WEBRTC_SPL_RSHIFT_U16(kLog10_2, 1));
-    tmp16no1 = WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio);
-    limiterLvl = targetLevelDbfs + tmp16no1;
-
-    // Calculate (through table lookup):
-    //  constMaxGain = log2(1+2^(log2(e)*diffGain)); (in Q8)
-    constMaxGain = kGenFuncTable[diffGain]; // in Q8
-
-    // Calculate a parameter used to approximate the fractional part of 2^x with a
-    // piecewise linear function in Q14:
-    //  constLinApprox = round(3/2*(4*(3-2*sqrt(2))/(log(2)^2)-0.5)*2^14);
-    constLinApprox = 22817; // in Q14
-
-    // Calculate a denominator used in the exponential part to convert from dB to linear scale:
-    //  den = 20*constMaxGain (in Q8)
-    den = WEBRTC_SPL_MUL_16_U16(20, constMaxGain); // in Q8
-
-    for (i = 0; i < 32; i++)
-    {
-        // Calculate scaled input level (compressor):
-        //  inLevel = fix((-constLog10_2*(compRatio-1)*(1-i)+fix(compRatio/2))/compRatio)
-        tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(kCompRatio - 1, i - 1); // Q0
-        tmp32 = WEBRTC_SPL_MUL_16_U16(tmp16, kLog10_2) + 1; // Q14
-        inLevel = WebRtcSpl_DivW32W16(tmp32, kCompRatio); // Q14
-
-        // Calculate diffGain-inLevel, to map using the genFuncTable
-        inLevel = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)diffGain, 14) - inLevel; // Q14
-
-        // Make calculations on abs(inLevel) and compensate for the sign afterwards.
-        absInLevel = (WebRtc_UWord32)WEBRTC_SPL_ABS_W32(inLevel); // Q14
-
-        // LUT with interpolation
-        intPart = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(absInLevel, 14);
-        fracPart = (WebRtc_UWord16)(absInLevel & 0x00003FFF); // extract the fractional part
-        tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8
-        tmpU32no1 = WEBRTC_SPL_UMUL_16_16(tmpU16, fracPart); // Q22
-        tmpU32no1 += WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)kGenFuncTable[intPart], 14); // Q22
-        logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 8); // Q14
-        // Compensate for negative exponent using the relation:
-        //  log2(1 + 2^-x) = log2(1 + 2^x) - x
-        if (inLevel < 0)
-        {
-            zeros = WebRtcSpl_NormU32(absInLevel);
-            zerosScale = 0;
-            if (zeros < 15)
-            {
-                // Not enough space for multiplication
-                tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(absInLevel, 15 - zeros); // Q(zeros-1)
-                tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no2, kLogE_1); // Q(zeros+13)
-                if (zeros < 9)
-                {
-                    tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 9 - zeros); // Q(zeros+13)
-                    zerosScale = 9 - zeros;
-                } else
-                {
-                    tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, zeros - 9); // Q22
-                }
-            } else
-            {
-                tmpU32no2 = WEBRTC_SPL_UMUL_32_16(absInLevel, kLogE_1); // Q28
-                tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6); // Q22
-            }
-            logApprox = 0;
-            if (tmpU32no2 < tmpU32no1)
-            {
-                logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1 - tmpU32no2, 8 - zerosScale); //Q14
-            }
-        }
-        numFIX = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_U16(maxGain, constMaxGain), 6); // Q14
-        numFIX -= WEBRTC_SPL_MUL_32_16((WebRtc_Word32)logApprox, diffGain); // Q14
-
-        // Calculate ratio
-        // Shift numFIX as much as possible
-        zeros = WebRtcSpl_NormW32(numFIX);
-        numFIX = WEBRTC_SPL_LSHIFT_W32(numFIX, zeros); // Q(14+zeros)
-
-        // Shift den so we end up in Qy1
-        tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 8); // Q(zeros)
-        if (numFIX < 0)
-        {
-            numFIX -= WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
-        } else
-        {
-            numFIX += WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
-        }
-        y32 = WEBRTC_SPL_DIV(numFIX, tmp32no1); // in Q14
-        if (limiterEnable && (i < limiterIdx))
-        {
-            tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14
-            tmp32 -= WEBRTC_SPL_LSHIFT_W32(limiterLvl, 14); // Q14
-            y32 = WebRtcSpl_DivW32W16(tmp32 + 10, 20);
-        }
-        if (y32 > 39000)
-        {
-            tmp32 = WEBRTC_SPL_MUL(y32 >> 1, kLog10) + 4096; // in Q27
-            tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 13); // in Q14
-        } else
-        {
-            tmp32 = WEBRTC_SPL_MUL(y32, kLog10) + 8192; // in Q28
-            tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 14); // in Q14
-        }
-        tmp32 += WEBRTC_SPL_LSHIFT_W32(16, 14); // in Q14 (Make sure final output is in Q16)
-
-        // Calculate power
-        if (tmp32 > 0)
-        {
-            intPart = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 14);
-            fracPart = (WebRtc_UWord16)(tmp32 & 0x00003FFF); // in Q14
-            if (WEBRTC_SPL_RSHIFT_W32(fracPart, 13))
-            {
-                tmp16 = WEBRTC_SPL_LSHIFT_W16(2, 14) - constLinApprox;
-                tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - fracPart;
-                tmp32no2 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16);
-                tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
-                tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - tmp32no2;
-            } else
-            {
-                tmp16 = constLinApprox - WEBRTC_SPL_LSHIFT_W16(1, 14);
-                tmp32no2 = WEBRTC_SPL_MUL_32_16(fracPart, tmp16);
-                tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
-            }
-            fracPart = (WebRtc_UWord16)tmp32no2;
-            gainTable[i] = WEBRTC_SPL_LSHIFT_W32(1, intPart)
-                    + WEBRTC_SPL_SHIFT_W32(fracPart, intPart - 14);
-        } else
-        {
-            gainTable[i] = 0;
-        }
-    }
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAgc_InitDigital(DigitalAgc_t *stt, WebRtc_Word16 agcMode)
-{
-
-    if (agcMode == kAgcModeFixedDigital)
-    {
-        // start at minimum to find correct gain faster
-        stt->capacitorSlow = 0;
-    } else
-    {
-        // start out with 0 dB gain
-        stt->capacitorSlow = 134217728; // (WebRtc_Word32)(0.125f * 32768.0f * 32768.0f);
-    }
-    stt->capacitorFast = 0;
-    stt->gain = 65536;
-    stt->gatePrevious = 0;
-    stt->agcMode = agcMode;
-#ifdef AGC_DEBUG
-    stt->frameCounter = 0;
-#endif
-
-    // initialize VADs
-    WebRtcAgc_InitVad(&stt->vadNearend);
-    WebRtcAgc_InitVad(&stt->vadFarend);
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAgc_AddFarendToDigital(DigitalAgc_t *stt, const WebRtc_Word16 *in_far,
-                                           WebRtc_Word16 nrSamples)
-{
-    // Check for valid pointer
-    if (&stt->vadFarend == NULL)
-    {
-        return -1;
-    }
-
-    // VAD for far end
-    WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples);
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const WebRtc_Word16 *in_near,
-                                       const WebRtc_Word16 *in_near_H, WebRtc_Word16 *out,
-                                       WebRtc_Word16 *out_H, WebRtc_UWord32 FS,
-                                       WebRtc_Word16 lowlevelSignal)
-{
-    // array for gains (one value per ms, incl start & end)
-    WebRtc_Word32 gains[11];
-
-    WebRtc_Word32 out_tmp, tmp32;
-    WebRtc_Word32 env[10];
-    WebRtc_Word32 nrg, max_nrg;
-    WebRtc_Word32 cur_level;
-    WebRtc_Word32 gain32, delta;
-    WebRtc_Word16 logratio;
-    WebRtc_Word16 lower_thr, upper_thr;
-    WebRtc_Word16 zeros, zeros_fast, frac;
-    WebRtc_Word16 decay;
-    WebRtc_Word16 gate, gain_adj;
-    WebRtc_Word16 k, n;
-    WebRtc_Word16 L, L2; // samples/subframe
-
-    // determine number of samples per ms
-    if (FS == 8000)
-    {
-        L = 8;
-        L2 = 3;
-    } else if (FS == 16000)
-    {
-        L = 16;
-        L2 = 4;
-    } else if (FS == 32000)
-    {
-        L = 16;
-        L2 = 4;
-    } else
-    {
-        return -1;
-    }
-
-    // TODO(andrew): again, we don't need input and output pointers...
-    if (in_near != out)
-    {
-        // Only needed if they don't already point to the same place.
-        memcpy(out, in_near, 10 * L * sizeof(WebRtc_Word16));
-    }
-    if (FS == 32000)
-    {
-        if (in_near_H != out_H)
-        {
-            memcpy(out_H, in_near_H, 10 * L * sizeof(WebRtc_Word16));
-        }
-    }
-    // VAD for near end
-    logratio = WebRtcAgc_ProcessVad(&stt->vadNearend, out, L * 10);
-
-    // Account for far end VAD
-    if (stt->vadFarend.counter > 10)
-    {
-        tmp32 = WEBRTC_SPL_MUL_16_16(3, logratio);
-        logratio = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 - stt->vadFarend.logRatio, 2);
-    }
-
-    // Determine decay factor depending on VAD
-    //  upper_thr = 1.0f;
-    //  lower_thr = 0.25f;
-    upper_thr = 1024; // Q10
-    lower_thr = 0; // Q10
-    if (logratio > upper_thr)
-    {
-        // decay = -2^17 / DecayTime;  ->  -65
-        decay = -65;
-    } else if (logratio < lower_thr)
-    {
-        decay = 0;
-    } else
-    {
-        // decay = (WebRtc_Word16)(((lower_thr - logratio)
-        //       * (2^27/(DecayTime*(upper_thr-lower_thr)))) >> 10);
-        // SUBSTITUTED: 2^27/(DecayTime*(upper_thr-lower_thr))  ->  65
-        tmp32 = WEBRTC_SPL_MUL_16_16((lower_thr - logratio), 65);
-        decay = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 10);
-    }
-
-    // adjust decay factor for long silence (detected as low standard deviation)
-    // This is only done in the adaptive modes
-    if (stt->agcMode != kAgcModeFixedDigital)
-    {
-        if (stt->vadNearend.stdLongTerm < 4000)
-        {
-            decay = 0;
-        } else if (stt->vadNearend.stdLongTerm < 8096)
-        {
-            // decay = (WebRtc_Word16)(((stt->vadNearend.stdLongTerm - 4000) * decay) >> 12);
-            tmp32 = WEBRTC_SPL_MUL_16_16((stt->vadNearend.stdLongTerm - 4000), decay);
-            decay = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
-        }
-
-        if (lowlevelSignal != 0)
-        {
-            decay = 0;
-        }
-    }
-#ifdef AGC_DEBUG
-    stt->frameCounter++;
-    fprintf(stt->logFile, "%5.2f\t%d\t%d\t%d\t", (float)(stt->frameCounter) / 100, logratio, decay, stt->vadNearend.stdLongTerm);
-#endif
-    // Find max amplitude per sub frame
-    // iterate over sub frames
-    for (k = 0; k < 10; k++)
-    {
-        // iterate over samples
-        max_nrg = 0;
-        for (n = 0; n < L; n++)
-        {
-            nrg = WEBRTC_SPL_MUL_16_16(out[k * L + n], out[k * L + n]);
-            if (nrg > max_nrg)
-            {
-                max_nrg = nrg;
-            }
-        }
-        env[k] = max_nrg;
-    }
-
-    // Calculate gain per sub frame
-    gains[0] = stt->gain;
-    for (k = 0; k < 10; k++)
-    {
-        // Fast envelope follower
-        //  decay time = -131000 / -1000 = 131 (ms)
-        stt->capacitorFast = AGC_SCALEDIFF32(-1000, stt->capacitorFast, stt->capacitorFast);
-        if (env[k] > stt->capacitorFast)
-        {
-            stt->capacitorFast = env[k];
-        }
-        // Slow envelope follower
-        if (env[k] > stt->capacitorSlow)
-        {
-            // increase capacitorSlow
-            stt->capacitorSlow
-                    = AGC_SCALEDIFF32(500, (env[k] - stt->capacitorSlow), stt->capacitorSlow);
-        } else
-        {
-            // decrease capacitorSlow
-            stt->capacitorSlow
-                    = AGC_SCALEDIFF32(decay, stt->capacitorSlow, stt->capacitorSlow);
-        }
-
-        // use maximum of both capacitors as current level
-        if (stt->capacitorFast > stt->capacitorSlow)
-        {
-            cur_level = stt->capacitorFast;
-        } else
-        {
-            cur_level = stt->capacitorSlow;
-        }
-        // Translate signal level into gain, using a piecewise linear approximation
-        // find number of leading zeros
-        zeros = WebRtcSpl_NormU32((WebRtc_UWord32)cur_level);
-        if (cur_level == 0)
-        {
-            zeros = 31;
-        }
-        tmp32 = (WEBRTC_SPL_LSHIFT_W32(cur_level, zeros) & 0x7FFFFFFF);
-        frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 19); // Q12
-        tmp32 = WEBRTC_SPL_MUL((stt->gainTable[zeros-1] - stt->gainTable[zeros]), frac);
-        gains[k + 1] = stt->gainTable[zeros] + WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
-#ifdef AGC_DEBUG
-        if (k == 0)
-        {
-            fprintf(stt->logFile, "%d\t%d\t%d\t%d\t%d\n", env[0], cur_level, stt->capacitorFast, stt->capacitorSlow, zeros);
-        }
-#endif
-    }
-
-    // Gate processing (lower gain during absence of speech)
-    zeros = WEBRTC_SPL_LSHIFT_W16(zeros, 9) - WEBRTC_SPL_RSHIFT_W16(frac, 3);
-    // find number of leading zeros
-    zeros_fast = WebRtcSpl_NormU32((WebRtc_UWord32)stt->capacitorFast);
-    if (stt->capacitorFast == 0)
-    {
-        zeros_fast = 31;
-    }
-    tmp32 = (WEBRTC_SPL_LSHIFT_W32(stt->capacitorFast, zeros_fast) & 0x7FFFFFFF);
-    zeros_fast = WEBRTC_SPL_LSHIFT_W16(zeros_fast, 9);
-    zeros_fast -= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 22);
-
-    gate = 1000 + zeros_fast - zeros - stt->vadNearend.stdShortTerm;
-
-    if (gate < 0)
-    {
-        stt->gatePrevious = 0;
-    } else
-    {
-        tmp32 = WEBRTC_SPL_MUL_16_16(stt->gatePrevious, 7);
-        gate = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)gate + tmp32, 3);
-        stt->gatePrevious = gate;
-    }
-    // gate < 0     -> no gate
-    // gate > 2500  -> max gate
-    if (gate > 0)
-    {
-        if (gate < 2500)
-        {
-            gain_adj = WEBRTC_SPL_RSHIFT_W16(2500 - gate, 5);
-        } else
-        {
-            gain_adj = 0;
-        }
-        for (k = 0; k < 10; k++)
-        {
-            if ((gains[k + 1] - stt->gainTable[0]) > 8388608)
-            {
-                // To prevent wraparound
-                tmp32 = WEBRTC_SPL_RSHIFT_W32((gains[k+1] - stt->gainTable[0]), 8);
-                tmp32 = WEBRTC_SPL_MUL(tmp32, (178 + gain_adj));
-            } else
-            {
-                tmp32 = WEBRTC_SPL_MUL((gains[k+1] - stt->gainTable[0]), (178 + gain_adj));
-                tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 8);
-            }
-            gains[k + 1] = stt->gainTable[0] + tmp32;
-        }
-    }
-
-    // Limit gain to avoid overload distortion
-    for (k = 0; k < 10; k++)
-    {
-        // To prevent wrap around
-        zeros = 10;
-        if (gains[k + 1] > 47453132)
-        {
-            zeros = 16 - WebRtcSpl_NormW32(gains[k + 1]);
-        }
-        gain32 = WEBRTC_SPL_RSHIFT_W32(gains[k+1], zeros) + 1;
-        gain32 = WEBRTC_SPL_MUL(gain32, gain32);
-        // check for overflow
-        while (AGC_MUL32(WEBRTC_SPL_RSHIFT_W32(env[k], 12) + 1, gain32)
-                > WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)32767, 2 * (1 - zeros + 10)))
-        {
-            // multiply by 253/256 ==> -0.1 dB
-            if (gains[k + 1] > 8388607)
-            {
-                // Prevent wrap around
-                gains[k + 1] = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(gains[k+1], 8), 253);
-            } else
-            {
-                gains[k + 1] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(gains[k+1], 253), 8);
-            }
-            gain32 = WEBRTC_SPL_RSHIFT_W32(gains[k+1], zeros) + 1;
-            gain32 = WEBRTC_SPL_MUL(gain32, gain32);
-        }
-    }
-    // gain reductions should be done 1 ms earlier than gain increases
-    for (k = 1; k < 10; k++)
-    {
-        if (gains[k] > gains[k + 1])
-        {
-            gains[k] = gains[k + 1];
-        }
-    }
-    // save start gain for next frame
-    stt->gain = gains[10];
-
-    // Apply gain
-    // handle first sub frame separately
-    delta = WEBRTC_SPL_LSHIFT_W32(gains[1] - gains[0], (4 - L2));
-    gain32 = WEBRTC_SPL_LSHIFT_W32(gains[0], 4);
-    // iterate over samples
-    for (n = 0; n < L; n++)
-    {
-        // For lower band
-        tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[n], WEBRTC_SPL_RSHIFT_W32(gain32 + 127, 7));
-        out_tmp = WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-        if (out_tmp > 4095)
-        {
-            out[n] = (WebRtc_Word16)32767;
-        } else if (out_tmp < -4096)
-        {
-            out[n] = (WebRtc_Word16)-32768;
-        } else
-        {
-            tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[n], WEBRTC_SPL_RSHIFT_W32(gain32, 4));
-            out[n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-        }
-        // For higher band
-        if (FS == 32000)
-        {
-            tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[n],
-                                   WEBRTC_SPL_RSHIFT_W32(gain32 + 127, 7));
-            out_tmp = WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-            if (out_tmp > 4095)
-            {
-                out_H[n] = (WebRtc_Word16)32767;
-            } else if (out_tmp < -4096)
-            {
-                out_H[n] = (WebRtc_Word16)-32768;
-            } else
-            {
-                tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[n],
-                                       WEBRTC_SPL_RSHIFT_W32(gain32, 4));
-                out_H[n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-            }
-        }
-        //
-
-        gain32 += delta;
-    }
-    // iterate over subframes
-    for (k = 1; k < 10; k++)
-    {
-        delta = WEBRTC_SPL_LSHIFT_W32(gains[k+1] - gains[k], (4 - L2));
-        gain32 = WEBRTC_SPL_LSHIFT_W32(gains[k], 4);
-        // iterate over samples
-        for (n = 0; n < L; n++)
-        {
-            // For lower band
-            tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[k * L + n],
-                                   WEBRTC_SPL_RSHIFT_W32(gain32, 4));
-            out[k * L + n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-            // For higher band
-            if (FS == 32000)
-            {
-                tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[k * L + n],
-                                       WEBRTC_SPL_RSHIFT_W32(gain32, 4));
-                out_H[k * L + n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-            }
-            gain32 += delta;
-        }
-    }
-
-    return 0;
-}
-
-void WebRtcAgc_InitVad(AgcVad_t *state)
-{
-    WebRtc_Word16 k;
-
-    state->HPstate = 0; // state of high pass filter
-    state->logRatio = 0; // log( P(active) / P(inactive) )
-    // average input level (Q10)
-    state->meanLongTerm = WEBRTC_SPL_LSHIFT_W16(15, 10);
-
-    // variance of input level (Q8)
-    state->varianceLongTerm = WEBRTC_SPL_LSHIFT_W32(500, 8);
-
-    state->stdLongTerm = 0; // standard deviation of input level in dB
-    // short-term average input level (Q10)
-    state->meanShortTerm = WEBRTC_SPL_LSHIFT_W16(15, 10);
-
-    // short-term variance of input level (Q8)
-    state->varianceShortTerm = WEBRTC_SPL_LSHIFT_W32(500, 8);
-
-    state->stdShortTerm = 0; // short-term standard deviation of input level in dB
-    state->counter = 3; // counts updates
-    for (k = 0; k < 8; k++)
-    {
-        // downsampling filter
-        state->downState[k] = 0;
-    }
-}
-
-WebRtc_Word16 WebRtcAgc_ProcessVad(AgcVad_t *state, // (i) VAD state
-                                   const WebRtc_Word16 *in, // (i) Speech signal
-                                   WebRtc_Word16 nrSamples) // (i) number of samples
-{
-    WebRtc_Word32 out, nrg, tmp32, tmp32b;
-    WebRtc_UWord16 tmpU16;
-    WebRtc_Word16 k, subfr, tmp16;
-    WebRtc_Word16 buf1[8];
-    WebRtc_Word16 buf2[4];
-    WebRtc_Word16 HPstate;
-    WebRtc_Word16 zeros, dB;
-
-    // process in 10 sub frames of 1 ms (to save on memory)
-    nrg = 0;
-    HPstate = state->HPstate;
-    for (subfr = 0; subfr < 10; subfr++)
-    {
-        // downsample to 4 kHz
-        if (nrSamples == 160)
-        {
-            for (k = 0; k < 8; k++)
-            {
-                tmp32 = (WebRtc_Word32)in[2 * k] + (WebRtc_Word32)in[2 * k + 1];
-                tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
-                buf1[k] = (WebRtc_Word16)tmp32;
-            }
-            in += 16;
-
-            WebRtcSpl_DownsampleBy2(buf1, 8, buf2, state->downState);
-        } else
-        {
-            WebRtcSpl_DownsampleBy2(in, 8, buf2, state->downState);
-            in += 8;
-        }
-
-        // high pass filter and compute energy
-        for (k = 0; k < 4; k++)
-        {
-            out = buf2[k] + HPstate;
-            tmp32 = WEBRTC_SPL_MUL(600, out);
-            HPstate = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(tmp32, 10) - buf2[k]);
-            tmp32 = WEBRTC_SPL_MUL(out, out);
-            nrg += WEBRTC_SPL_RSHIFT_W32(tmp32, 6);
-        }
-    }
-    state->HPstate = HPstate;
-
-    // find number of leading zeros
-    if (!(0xFFFF0000 & nrg))
-    {
-        zeros = 16;
-    } else
-    {
-        zeros = 0;
-    }
-    if (!(0xFF000000 & (nrg << zeros)))
-    {
-        zeros += 8;
-    }
-    if (!(0xF0000000 & (nrg << zeros)))
-    {
-        zeros += 4;
-    }
-    if (!(0xC0000000 & (nrg << zeros)))
-    {
-        zeros += 2;
-    }
-    if (!(0x80000000 & (nrg << zeros)))
-    {
-        zeros += 1;
-    }
-
-    // energy level (range {-32..30}) (Q10)
-    dB = WEBRTC_SPL_LSHIFT_W16(15 - zeros, 11);
-
-    // Update statistics
-
-    if (state->counter < kAvgDecayTime)
-    {
-        // decay time = AvgDecTime * 10 ms
-        state->counter++;
-    }
-
-    // update short-term estimate of mean energy level (Q10)
-    tmp32 = (WEBRTC_SPL_MUL_16_16(state->meanShortTerm, 15) + (WebRtc_Word32)dB);
-    state->meanShortTerm = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
-
-    // update short-term estimate of variance in energy level (Q8)
-    tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(dB, dB), 12);
-    tmp32 += WEBRTC_SPL_MUL(state->varianceShortTerm, 15);
-    state->varianceShortTerm = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
-
-    // update short-term estimate of standard deviation in energy level (Q10)
-    tmp32 = WEBRTC_SPL_MUL_16_16(state->meanShortTerm, state->meanShortTerm);
-    tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceShortTerm, 12) - tmp32;
-    state->stdShortTerm = (WebRtc_Word16)WebRtcSpl_Sqrt(tmp32);
-
-    // update long-term estimate of mean energy level (Q10)
-    tmp32 = WEBRTC_SPL_MUL_16_16(state->meanLongTerm, state->counter) + (WebRtc_Word32)dB;
-    state->meanLongTerm = WebRtcSpl_DivW32W16ResW16(tmp32,
-                                                    WEBRTC_SPL_ADD_SAT_W16(state->counter, 1));
-
-    // update long-term estimate of variance in energy level (Q8)
-    tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(dB, dB), 12);
-    tmp32 += WEBRTC_SPL_MUL(state->varianceLongTerm, state->counter);
-    state->varianceLongTerm = WebRtcSpl_DivW32W16(tmp32,
-                                                  WEBRTC_SPL_ADD_SAT_W16(state->counter, 1));
-
-    // update long-term estimate of standard deviation in energy level (Q10)
-    tmp32 = WEBRTC_SPL_MUL_16_16(state->meanLongTerm, state->meanLongTerm);
-    tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceLongTerm, 12) - tmp32;
-    state->stdLongTerm = (WebRtc_Word16)WebRtcSpl_Sqrt(tmp32);
-
-    // update voice activity measure (Q10)
-    tmp16 = WEBRTC_SPL_LSHIFT_W16(3, 12);
-    tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, (dB - state->meanLongTerm));
-    tmp32 = WebRtcSpl_DivW32W16(tmp32, state->stdLongTerm);
-    tmpU16 = WEBRTC_SPL_LSHIFT_U16((WebRtc_UWord16)13, 12);
-    tmp32b = WEBRTC_SPL_MUL_16_U16(state->logRatio, tmpU16);
-    tmp32 += WEBRTC_SPL_RSHIFT_W32(tmp32b, 10);
-
-    state->logRatio = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 6);
-
-    // limit
-    if (state->logRatio > 2048)
-    {
-        state->logRatio = 2048;
-    }
-    if (state->logRatio < -2048)
-    {
-        state->logRatio = -2048;
-    }
-
-    return state->logRatio; // Q10
-}

src/modules/audio_processing/agc/digital_agc.h

@@ -1,76 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
-
-#ifdef AGC_DEBUG
-#include <stdio.h>
-#endif
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-// the 32 most significant bits of A(19) * B(26) >> 13
-#define AGC_MUL32(A, B)             (((B)>>13)*(A) + ( ((0x00001FFF & (B))*(A)) >> 13 ))
-// C + the 32 most significant bits of A * B
-#define AGC_SCALEDIFF32(A, B, C)    ((C) + ((B)>>16)*(A) + ( ((0x0000FFFF & (B))*(A)) >> 16 ))
-
-typedef struct
-{
-    WebRtc_Word32 downState[8];
-    WebRtc_Word16 HPstate;
-    WebRtc_Word16 counter;
-    WebRtc_Word16 logRatio; // log( P(active) / P(inactive) ) (Q10)
-    WebRtc_Word16 meanLongTerm; // Q10
-    WebRtc_Word32 varianceLongTerm; // Q8
-    WebRtc_Word16 stdLongTerm; // Q10
-    WebRtc_Word16 meanShortTerm; // Q10
-    WebRtc_Word32 varianceShortTerm; // Q8
-    WebRtc_Word16 stdShortTerm; // Q10
-} AgcVad_t; // total = 54 bytes
-
-typedef struct
-{
-    WebRtc_Word32 capacitorSlow;
-    WebRtc_Word32 capacitorFast;
-    WebRtc_Word32 gain;
-    WebRtc_Word32 gainTable[32];
-    WebRtc_Word16 gatePrevious;
-    WebRtc_Word16 agcMode;
-    AgcVad_t      vadNearend;
-    AgcVad_t      vadFarend;
-#ifdef AGC_DEBUG
-    FILE*         logFile;
-    int           frameCounter;
-#endif
-} DigitalAgc_t;
-
-WebRtc_Word32 WebRtcAgc_InitDigital(DigitalAgc_t *digitalAgcInst, WebRtc_Word16 agcMode);
-
-WebRtc_Word32 WebRtcAgc_ProcessDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inNear,
-                             const WebRtc_Word16 *inNear_H, WebRtc_Word16 *out,
-                             WebRtc_Word16 *out_H, WebRtc_UWord32 FS,
-                             WebRtc_Word16 lowLevelSignal);
-
-WebRtc_Word32 WebRtcAgc_AddFarendToDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inFar,
-                                 WebRtc_Word16 nrSamples);
-
-void WebRtcAgc_InitVad(AgcVad_t *vadInst);
-
-WebRtc_Word16 WebRtcAgc_ProcessVad(AgcVad_t *vadInst, // (i) VAD state
-                            const WebRtc_Word16 *in, // (i) Speech signal
-                            WebRtc_Word16 nrSamples); // (i) number of samples
-
-WebRtc_Word32 WebRtcAgc_CalculateGainTable(WebRtc_Word32 *gainTable, // Q16
-                                 WebRtc_Word16 compressionGaindB, // Q0 (in dB)
-                                 WebRtc_Word16 targetLevelDbfs,// Q0 (in dB)
-                                 WebRtc_UWord8 limiterEnable, WebRtc_Word16 analogTarget);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_

src/modules/audio_processing/agc/interface/gain_control.h

@@ -1,273 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_INTERFACE_GAIN_CONTROL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_INTERFACE_GAIN_CONTROL_H_
-
-#include "typedefs.h"
-
-// Errors
-#define AGC_UNSPECIFIED_ERROR           18000
-#define AGC_UNSUPPORTED_FUNCTION_ERROR  18001
-#define AGC_UNINITIALIZED_ERROR         18002
-#define AGC_NULL_POINTER_ERROR          18003
-#define AGC_BAD_PARAMETER_ERROR         18004
-
-// Warnings
-#define AGC_BAD_PARAMETER_WARNING       18050
-
-enum
-{
-    kAgcModeUnchanged,
-    kAgcModeAdaptiveAnalog,
-    kAgcModeAdaptiveDigital,
-    kAgcModeFixedDigital
-};
-
-enum
-{
-    kAgcFalse = 0,
-    kAgcTrue
-};
-
-typedef struct
-{
-    WebRtc_Word16 targetLevelDbfs;   // default 3 (-3 dBOv)
-    WebRtc_Word16 compressionGaindB; // default 9 dB
-    WebRtc_UWord8 limiterEnable;     // default kAgcTrue (on)
-} WebRtcAgc_config_t;
-
-#if defined(__cplusplus)
-extern "C"
-{
-#endif
-
-/*
- * This function processes a 10/20ms frame of far-end speech to determine
- * if there is active speech. Far-end speech length can be either 10ms or
- * 20ms. The length of the input speech vector must be given in samples
- * (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000).
- *
- * Input:
- *      - agcInst           : AGC instance.
- *      - inFar             : Far-end input speech vector (10 or 20ms)
- *      - samples           : Number of samples in input vector
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_AddFarend(void* agcInst,
-                        const WebRtc_Word16* inFar,
-                        WebRtc_Word16 samples);
-
-/*
- * This function processes a 10/20ms frame of microphone speech to determine
- * if there is active speech. Microphone speech length can be either 10ms or
- * 20ms. The length of the input speech vector must be given in samples
- * (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000). For very low
- * input levels, the input signal is increased in level by multiplying and
- * overwriting the samples in inMic[].
- *
- * This function should be called before any further processing of the
- * near-end microphone signal.
- *
- * Input:
- *      - agcInst           : AGC instance.
- *      - inMic             : Microphone input speech vector (10 or 20 ms) for
- *                            L band
- *      - inMic_H           : Microphone input speech vector (10 or 20 ms) for
- *                            H band
- *      - samples           : Number of samples in input vector
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_AddMic(void* agcInst,
-                     WebRtc_Word16* inMic,
-                     WebRtc_Word16* inMic_H,
-                     WebRtc_Word16 samples);
-
-/*
- * This function replaces the analog microphone with a virtual one.
- * It is a digital gain applied to the input signal and is used in the
- * agcAdaptiveDigital mode where no microphone level is adjustable.
- * Microphone speech length can be either 10ms or 20ms. The length of the
- * input speech vector must be given in samples (80/160 when FS=8000, and
- * 160/320 when FS=16000 or FS=32000).
- *
- * Input:
- *      - agcInst           : AGC instance.
- *      - inMic             : Microphone input speech vector for (10 or 20 ms)
- *                            L band
- *      - inMic_H           : Microphone input speech vector for (10 or 20 ms)
- *                            H band
- *      - samples           : Number of samples in input vector
- *      - micLevelIn        : Input level of microphone (static)
- *
- * Output:
- *      - inMic             : Microphone output after processing (L band)
- *      - inMic_H           : Microphone output after processing (H band)
- *      - micLevelOut       : Adjusted microphone level after processing
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_VirtualMic(void* agcInst,
-                         WebRtc_Word16* inMic,
-                         WebRtc_Word16* inMic_H,
-                         WebRtc_Word16 samples,
-                         WebRtc_Word32 micLevelIn,
-                         WebRtc_Word32* micLevelOut);
-
-/*
- * This function processes a 10/20ms frame and adjusts (normalizes) the gain
- * both analog and digitally. The gain adjustments are done only during
- * active periods of speech. The input speech length can be either 10ms or
- * 20ms and the output is of the same length. The length of the speech
- * vectors must be given in samples (80/160 when FS=8000, and 160/320 when
- * FS=16000 or FS=32000). The echo parameter can be used to ensure the AGC will
- * not adjust upward in the presence of echo.
- *
- * This function should be called after processing the near-end microphone
- * signal, in any case after any echo cancellation.
- *
- * Input:
- *      - agcInst           : AGC instance
- *      - inNear            : Near-end input speech vector (10 or 20 ms) for
- *                            L band
- *      - inNear_H          : Near-end input speech vector (10 or 20 ms) for
- *                            H band
- *      - samples           : Number of samples in input/output vector
- *      - inMicLevel        : Current microphone volume level
- *      - echo              : Set to 0 if the signal passed to add_mic is
- *                            almost certainly free of echo; otherwise set
- *                            to 1. If you have no information regarding echo
- *                            set to 0.
- *
- * Output:
- *      - outMicLevel       : Adjusted microphone volume level
- *      - out               : Gain-adjusted near-end speech vector (L band)
- *                          : May be the same vector as the input.
- *      - out_H             : Gain-adjusted near-end speech vector (H band)
- *      - saturationWarning : A returned value of 1 indicates a saturation event
- *                            has occurred and the volume cannot be further
- *                            reduced. Otherwise will be set to 0.
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_Process(void* agcInst,
-                      const WebRtc_Word16* inNear,
-                      const WebRtc_Word16* inNear_H,
-                      WebRtc_Word16 samples,
-                      WebRtc_Word16* out,
-                      WebRtc_Word16* out_H,
-                      WebRtc_Word32 inMicLevel,
-                      WebRtc_Word32* outMicLevel,
-                      WebRtc_Word16 echo,
-                      WebRtc_UWord8* saturationWarning);
-
-/*
- * This function sets the config parameters (targetLevelDbfs,
- * compressionGaindB and limiterEnable).
- *
- * Input:
- *      - agcInst           : AGC instance
- *      - config            : config struct
- *
- * Output:
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_set_config(void* agcInst, WebRtcAgc_config_t config);
-
-/*
- * This function returns the config parameters (targetLevelDbfs,
- * compressionGaindB and limiterEnable).
- *
- * Input:
- *      - agcInst           : AGC instance
- *
- * Output:
- *      - config            : config struct
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_get_config(void* agcInst, WebRtcAgc_config_t* config);
-
-/*
- * This function creates an AGC instance, which will contain the state
- * information for one (duplex) channel.
- *
- * Return value             : AGC instance if successful
- *                          : 0 (i.e., a NULL pointer) if unsuccessful
- */
-int WebRtcAgc_Create(void **agcInst);
-
-/*
- * This function frees the AGC instance created at the beginning.
- *
- * Input:
- *      - agcInst           : AGC instance.
- *
- * Return value             :  0 - Ok
- *                            -1 - Error
- */
-int WebRtcAgc_Free(void *agcInst);
-
-/*
- * This function initializes an AGC instance.
- *
- * Input:
- *      - agcInst           : AGC instance.
- *      - minLevel          : Minimum possible mic level
- *      - maxLevel          : Maximum possible mic level
- *      - agcMode           : 0 - Unchanged
- *                          : 1 - Adaptive Analog Automatic Gain Control -3dBOv
- *                          : 2 - Adaptive Digital Automatic Gain Control -3dBOv
- *                          : 3 - Fixed Digital Gain 0dB
- *      - fs                : Sampling frequency
- *
- * Return value             :  0 - Ok
- *                            -1 - Error
- */
-int WebRtcAgc_Init(void *agcInst,
-                   WebRtc_Word32 minLevel,
-                   WebRtc_Word32 maxLevel,
-                   WebRtc_Word16 agcMode,
-                   WebRtc_UWord32 fs);
-
-/*
- * This function returns a text string containing the version.
- *
- * Input:
- *      - length            : Length of the char array pointed to by version
- * Output:
- *      - version           : Pointer to a char array of to which the version
- *                          : string will be copied.
- *
- * Return value             :  0 - OK
- *                            -1 - Error
- */
-int WebRtcAgc_Version(WebRtc_Word8 *versionStr, WebRtc_Word16 length);
-
-#if defined(__cplusplus)
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_INTERFACE_GAIN_CONTROL_H_

src/modules/audio_processing/apm_tests.gypi

@@ -1,100 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'variables': {
-    'protoc_out_dir': '<(SHARED_INTERMEDIATE_DIR)/protoc_out',
-    'protoc_out_relpath': 'webrtc/audio_processing',
-  },
-  'targets': [
-    {
-      'target_name': 'audioproc_unittest',
-      'type': 'executable',
-      'conditions': [
-        ['prefer_fixed_point==1', {
-          'defines': ['WEBRTC_APM_UNIT_TEST_FIXED_PROFILE'],
-        }, {
-          'defines': ['WEBRTC_APM_UNIT_TEST_FLOAT_PROFILE'],
-        }],
-      ],
-      'dependencies': [
-        'audioproc_unittest_proto',
-        'audio_processing',
-        '<(webrtc_root)/common_audio/common_audio.gyp:spl',
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-        '<(webrtc_root)/../test/test.gyp:test_support',
-        '<(webrtc_root)/../testing/gtest.gyp:gtest',
-        '<(webrtc_root)/../third_party/protobuf/protobuf.gyp:protobuf_lite',
-      ],
-      'include_dirs': [
-        '<(webrtc_root)/../testing/gtest/include',
-        '<(protoc_out_dir)',
-      ],
-      'sources': [
-        'test/unit_test.cc',
-        '<(protoc_out_dir)/<(protoc_out_relpath)/unittest.pb.cc',
-        '<(protoc_out_dir)/<(protoc_out_relpath)/unittest.pb.h',
-      ],
-    },
-    {
-      # Protobuf compiler / generate rule for audioproc_unittest
-      'target_name': 'audioproc_unittest_proto',
-      'type': 'none',
-      'variables': {
-        'proto_relpath':
-          '<(webrtc_root)/modules/audio_processing/test',
-      },
-      'sources': [
-        '<(proto_relpath)/unittest.proto',
-      ],
-      'rules': [
-        {
-          'rule_name': 'genproto',
-          'extension': 'proto',
-          'inputs': [
-            '<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)protoc<(EXECUTABLE_SUFFIX)',
-          ],
-          'outputs': [
-            '<(protoc_out_dir)/<(protoc_out_relpath)/<(RULE_INPUT_ROOT).pb.cc',
-            '<(protoc_out_dir)/<(RULE_INPUT_ROOT).pb.h',
-          ],
-          'action': [
-            '<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)protoc<(EXECUTABLE_SUFFIX)',
-            '--proto_path=<(proto_relpath)',
-            '<(proto_relpath)/<(RULE_INPUT_NAME)',
-            '--cpp_out=<(protoc_out_dir)/<(protoc_out_relpath)',
-          ],
-          'message': 'Generating C++ code from <(RULE_INPUT_PATH)',
-        },
-      ],
-      'dependencies': [
-        '<(webrtc_root)/../third_party/protobuf/protobuf.gyp:protoc#host',
-      ],
-      # This target exports a hard dependency because it generates header
-      # files.
-      'hard_dependency': 1,
-    },
-    {
-      'target_name': 'audioproc_process_test',
-      'type': 'executable',
-      'dependencies': [
-        'audio_processing',
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-        '<(webrtc_root)/../testing/gtest.gyp:gtest',
-        '<(webrtc_root)/../third_party/protobuf/protobuf.gyp:protobuf_lite',
-      ],
-      'include_dirs': [
-        '<(webrtc_root)/../testing/gtest/include',
-        '<(protoc_out_dir)',
-      ],
-      'sources': [
-        'test/process_test.cc',
-      ],
-    },
-  ],
-}

src/modules/audio_processing/audio_buffer.cc

@@ -1,288 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_buffer.h"
-
-namespace webrtc {
-namespace {
-
-enum {
-  kSamplesPer8kHzChannel = 80,
-  kSamplesPer16kHzChannel = 160,
-  kSamplesPer32kHzChannel = 320
-};
-
-void StereoToMono(const WebRtc_Word16* left, const WebRtc_Word16* right,
-                  WebRtc_Word16* out, int samples_per_channel) {
-  WebRtc_Word32 data_int32 = 0;
-  for (int i = 0; i < samples_per_channel; i++) {
-    data_int32 = (left[i] + right[i]) >> 1;
-    if (data_int32 > 32767) {
-      data_int32 = 32767;
-    } else if (data_int32 < -32768) {
-      data_int32 = -32768;
-    }
-
-    out[i] = static_cast<WebRtc_Word16>(data_int32);
-  }
-}
-}  // namespace
-
-struct AudioChannel {
-  AudioChannel() {
-    memset(data, 0, sizeof(data));
-  }
-
-  WebRtc_Word16 data[kSamplesPer32kHzChannel];
-};
-
-struct SplitAudioChannel {
-  SplitAudioChannel() {
-    memset(low_pass_data, 0, sizeof(low_pass_data));
-    memset(high_pass_data, 0, sizeof(high_pass_data));
-    memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1));
-    memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2));
-    memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1));
-    memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2));
-  }
-
-  WebRtc_Word16 low_pass_data[kSamplesPer16kHzChannel];
-  WebRtc_Word16 high_pass_data[kSamplesPer16kHzChannel];
-
-  WebRtc_Word32 analysis_filter_state1[6];
-  WebRtc_Word32 analysis_filter_state2[6];
-  WebRtc_Word32 synthesis_filter_state1[6];
-  WebRtc_Word32 synthesis_filter_state2[6];
-};
-
-// TODO(andrew): check range of input parameters?
-AudioBuffer::AudioBuffer(int max_num_channels,
-                         int samples_per_channel)
-  : max_num_channels_(max_num_channels),
-    num_channels_(0),
-    num_mixed_channels_(0),
-    num_mixed_low_pass_channels_(0),
-    samples_per_channel_(samples_per_channel),
-    samples_per_split_channel_(samples_per_channel),
-    reference_copied_(false),
-    activity_(AudioFrame::kVadUnknown),
-    data_(NULL),
-    channels_(NULL),
-    split_channels_(NULL),
-    mixed_low_pass_channels_(NULL),
-    low_pass_reference_channels_(NULL) {
-  if (max_num_channels_ > 1) {
-    channels_ = new AudioChannel[max_num_channels_];
-    mixed_low_pass_channels_ = new AudioChannel[max_num_channels_];
-  }
-  low_pass_reference_channels_ = new AudioChannel[max_num_channels_];
-
-  if (samples_per_channel_ == kSamplesPer32kHzChannel) {
-    split_channels_ = new SplitAudioChannel[max_num_channels_];
-    samples_per_split_channel_ = kSamplesPer16kHzChannel;
-  }
-}
-
-AudioBuffer::~AudioBuffer() {
-  if (channels_ != NULL) {
-    delete [] channels_;
-  }
-
-  if (mixed_low_pass_channels_ != NULL) {
-    delete [] mixed_low_pass_channels_;
-  }
-
-  if (low_pass_reference_channels_ != NULL) {
-    delete [] low_pass_reference_channels_;
-  }
-
-  if (split_channels_ != NULL) {
-    delete [] split_channels_;
-  }
-}
-
-WebRtc_Word16* AudioBuffer::data(int channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  if (data_ != NULL) {
-    return data_;
-  }
-
-  return channels_[channel].data;
-}
-
-WebRtc_Word16* AudioBuffer::low_pass_split_data(int channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  if (split_channels_ == NULL) {
-    return data(channel);
-  }
-
-  return split_channels_[channel].low_pass_data;
-}
-
-WebRtc_Word16* AudioBuffer::high_pass_split_data(int channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  if (split_channels_ == NULL) {
-    return NULL;
-  }
-
-  return split_channels_[channel].high_pass_data;
-}
-
-WebRtc_Word16* AudioBuffer::mixed_low_pass_data(int channel) const {
-  assert(channel >= 0 && channel < num_mixed_low_pass_channels_);
-
-  return mixed_low_pass_channels_[channel].data;
-}
-
-WebRtc_Word16* AudioBuffer::low_pass_reference(int channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  if (!reference_copied_) {
-    return NULL;
-  }
-
-  return low_pass_reference_channels_[channel].data;
-}
-
-WebRtc_Word32* AudioBuffer::analysis_filter_state1(int channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  return split_channels_[channel].analysis_filter_state1;
-}
-
-WebRtc_Word32* AudioBuffer::analysis_filter_state2(int channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  return split_channels_[channel].analysis_filter_state2;
-}
-
-WebRtc_Word32* AudioBuffer::synthesis_filter_state1(int channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  return split_channels_[channel].synthesis_filter_state1;
-}
-
-WebRtc_Word32* AudioBuffer::synthesis_filter_state2(int channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  return split_channels_[channel].synthesis_filter_state2;
-}
-
-void AudioBuffer::set_activity(AudioFrame::VADActivity activity) {
-  activity_ = activity;
-}
-
-AudioFrame::VADActivity AudioBuffer::activity() {
-  return activity_;
-}
-
-int AudioBuffer::num_channels() const {
-  return num_channels_;
-}
-
-int AudioBuffer::samples_per_channel() const {
-  return samples_per_channel_;
-}
-
-int AudioBuffer::samples_per_split_channel() const {
-  return samples_per_split_channel_;
-}
-
-// TODO(andrew): Do deinterleaving and mixing in one step?
-void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
-  assert(frame->_audioChannel <= max_num_channels_);
-  assert(frame->_payloadDataLengthInSamples ==  samples_per_channel_);
-
-  num_channels_ = frame->_audioChannel;
-  num_mixed_channels_ = 0;
-  num_mixed_low_pass_channels_ = 0;
-  reference_copied_ = false;
-  activity_ = frame->_vadActivity;
-
-  if (num_channels_ == 1) {
-    // We can get away with a pointer assignment in this case.
-    data_ = frame->_payloadData;
-    return;
-  }
-
-  WebRtc_Word16* interleaved = frame->_payloadData;
-  for (int i = 0; i < num_channels_; i++) {
-    WebRtc_Word16* deinterleaved = channels_[i].data;
-    int interleaved_idx = i;
-    for (int j = 0; j < samples_per_channel_; j++) {
-      deinterleaved[j] = interleaved[interleaved_idx];
-      interleaved_idx += num_channels_;
-    }
-  }
-}
-
-void AudioBuffer::InterleaveTo(AudioFrame* frame) const {
-  assert(frame->_audioChannel == num_channels_);
-  assert(frame->_payloadDataLengthInSamples == samples_per_channel_);
-  frame->_vadActivity = activity_;
-
-  if (num_channels_ == 1) {
-    if (num_mixed_channels_ == 1) {
-      memcpy(frame->_payloadData,
-             channels_[0].data,
-             sizeof(WebRtc_Word16) * samples_per_channel_);
-    } else {
-      // These should point to the same buffer in this case.
-      assert(data_ == frame->_payloadData);
-    }
-
-    return;
-  }
-
-  WebRtc_Word16* interleaved = frame->_payloadData;
-  for (int i = 0; i < num_channels_; i++) {
-    WebRtc_Word16* deinterleaved = channels_[i].data;
-    int interleaved_idx = i;
-    for (int j = 0; j < samples_per_channel_; j++) {
-      interleaved[interleaved_idx] = deinterleaved[j];
-      interleaved_idx += num_channels_;
-    }
-  }
-}
-
-// TODO(andrew): would be good to support the no-mix case with pointer
-// assignment.
-// TODO(andrew): handle mixing to multiple channels?
-void AudioBuffer::Mix(int num_mixed_channels) {
-  // We currently only support the stereo to mono case.
-  assert(num_channels_ == 2);
-  assert(num_mixed_channels == 1);
-
-  StereoToMono(channels_[0].data,
-               channels_[1].data,
-               channels_[0].data,
-               samples_per_channel_);
-
-  num_channels_ = num_mixed_channels;
-  num_mixed_channels_ = num_mixed_channels;
-}
-
-void AudioBuffer::CopyAndMixLowPass(int num_mixed_channels) {
-  // We currently only support the stereo to mono case.
-  assert(num_channels_ == 2);
-  assert(num_mixed_channels == 1);
-
-  StereoToMono(low_pass_split_data(0),
-               low_pass_split_data(1),
-               mixed_low_pass_channels_[0].data,
-               samples_per_split_channel_);
-
-  num_mixed_low_pass_channels_ = num_mixed_channels;
-}
-
-void AudioBuffer::CopyLowPassToReference() {
-  reference_copied_ = true;
-  for (int i = 0; i < num_channels_; i++) {
-    memcpy(low_pass_reference_channels_[i].data,
-           low_pass_split_data(i),
-           sizeof(WebRtc_Word16) * samples_per_split_channel_);
-  }
-}
-}  // namespace webrtc

src/modules/audio_processing/audio_buffer.h

@@ -1,71 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
-
-#include "module_common_types.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-struct AudioChannel;
-struct SplitAudioChannel;
-
-class AudioBuffer {
- public:
-  AudioBuffer(int max_num_channels, int samples_per_channel);
-  virtual ~AudioBuffer();
-
-  int num_channels() const;
-  int samples_per_channel() const;
-  int samples_per_split_channel() const;
-
-  WebRtc_Word16* data(int channel) const;
-  WebRtc_Word16* low_pass_split_data(int channel) const;
-  WebRtc_Word16* high_pass_split_data(int channel) const;
-  WebRtc_Word16* mixed_low_pass_data(int channel) const;
-  WebRtc_Word16* low_pass_reference(int channel) const;
-
-  WebRtc_Word32* analysis_filter_state1(int channel) const;
-  WebRtc_Word32* analysis_filter_state2(int channel) const;
-  WebRtc_Word32* synthesis_filter_state1(int channel) const;
-  WebRtc_Word32* synthesis_filter_state2(int channel) const;
-
-  void set_activity(AudioFrame::VADActivity activity);
-  AudioFrame::VADActivity activity();
-
-  void DeinterleaveFrom(AudioFrame* audioFrame);
-  void InterleaveTo(AudioFrame* audioFrame) const;
-  void Mix(int num_mixed_channels);
-  void CopyAndMixLowPass(int num_mixed_channels);
-  void CopyLowPassToReference();
-
- private:
-  const int max_num_channels_;
-  int num_channels_;
-  int num_mixed_channels_;
-  int num_mixed_low_pass_channels_;
-  const int samples_per_channel_;
-  int samples_per_split_channel_;
-  bool reference_copied_;
-  AudioFrame::VADActivity activity_;
-
-  WebRtc_Word16* data_;
-  // TODO(andrew): use vectors here.
-  AudioChannel* channels_;
-  SplitAudioChannel* split_channels_;
-  // TODO(andrew): improve this, we don't need the full 32 kHz space here.
-  AudioChannel* mixed_low_pass_channels_;
-  AudioChannel* low_pass_reference_channels_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_

src/modules/audio_processing/audio_processing.gypi

@@ -1,130 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'variables': {
-    'protoc_out_dir': '<(SHARED_INTERMEDIATE_DIR)/protoc_out',
-    'protoc_out_relpath': 'webrtc/audio_processing',
-  },
-  'targets': [
-    {
-      'target_name': 'audio_processing',
-      'type': '<(library)',
-      'conditions': [
-        ['prefer_fixed_point==1', {
-          'dependencies': ['ns_fix'],
-          'defines': ['WEBRTC_NS_FIXED'],
-        }, {
-          'dependencies': ['ns'],
-          'defines': ['WEBRTC_NS_FLOAT'],
-        }],
-        ['build_with_chromium==1', {
-          'dependencies': [
-            '<(webrtc_root)/../protobuf/protobuf.gyp:protobuf_lite',
-          ],
-        }, {
-          'dependencies': [
-            '<(webrtc_root)/../third_party/protobuf/protobuf.gyp:protobuf_lite',
-          ],
-        }],
-      ],
-      'dependencies': [
-        'debug_proto',
-        'aec',
-        'aecm',
-        'agc',
-        '<(webrtc_root)/common_audio/common_audio.gyp:spl',
-        '<(webrtc_root)/common_audio/common_audio.gyp:vad',
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-      ],
-      'include_dirs': [
-        'interface',
-        '../interface',
-        '<(protoc_out_dir)',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          'interface',
-          '../interface',
-        ],
-      },
-      'sources': [
-        'interface/audio_processing.h',
-        'audio_buffer.cc',
-        'audio_buffer.h',
-        'audio_processing_impl.cc',
-        'audio_processing_impl.h',
-        'echo_cancellation_impl.cc',
-        'echo_cancellation_impl.h',
-        'echo_control_mobile_impl.cc',
-        'echo_control_mobile_impl.h',
-        'gain_control_impl.cc',
-        'gain_control_impl.h',
-        'high_pass_filter_impl.cc',
-        'high_pass_filter_impl.h',
-        'level_estimator_impl.cc',
-        'level_estimator_impl.h',
-        'noise_suppression_impl.cc',
-        'noise_suppression_impl.h',
-        'splitting_filter.cc',
-        'splitting_filter.h',
-        'processing_component.cc',
-        'processing_component.h',
-        'voice_detection_impl.cc',
-        'voice_detection_impl.h',
-        '<(protoc_out_dir)/<(protoc_out_relpath)/debug.pb.cc',
-        '<(protoc_out_dir)/<(protoc_out_relpath)/debug.pb.h',
-      ],
-    },
-    {
-      # Protobuf compiler / generate rule for audio_processing
-      'target_name': 'debug_proto',
-      'type': 'none',
-      'variables': {
-        'proto_relpath': '<(webrtc_root)/modules/audio_processing',
-      },
-      'sources': [
-        '<(proto_relpath)/debug.proto',
-      ],
-      'rules': [
-        {
-          'rule_name': 'genproto',
-          'extension': 'proto',
-          'inputs': [
-            '<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)protoc<(EXECUTABLE_SUFFIX)',
-          ],
-          'outputs': [
-            '<(protoc_out_dir)/<(protoc_out_relpath)/<(RULE_INPUT_ROOT).pb.cc',
-            '<(protoc_out_dir)/<(protoc_out_relpath)/<(RULE_INPUT_ROOT).pb.h',
-          ],
-          'action': [
-            '<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)protoc<(EXECUTABLE_SUFFIX)',
-            '--proto_path=<(proto_relpath)',
-            '<(proto_relpath)/<(RULE_INPUT_NAME)',
-            '--cpp_out=<(protoc_out_dir)/<(protoc_out_relpath)',
-          ],
-          'message': 'Generating C++ code from <(RULE_INPUT_PATH)',
-        },
-      ],
-      'conditions': [
-        ['build_with_chromium==1', {
-          'dependencies': [
-            '<(webrtc_root)/../protobuf/protobuf.gyp:protoc#host',
-          ],
-        }, {
-          'dependencies': [
-            '<(webrtc_root)/../third_party/protobuf/protobuf.gyp:protoc#host',
-          ],
-        }],
-      ],
-      # This target exports a hard dependency because it generates header
-      # files.
-      'hard_dependency': 1,
-    },
-  ],
-}

src/modules/audio_processing/audio_processing_impl.cc

@@ -1,673 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_processing_impl.h"
-
-#include <assert.h>
-
-#include "audio_buffer.h"
-#include "critical_section_wrapper.h"
-#include "echo_cancellation_impl.h"
-#include "echo_control_mobile_impl.h"
-#ifndef NDEBUG
-#include "file_wrapper.h"
-#endif
-#include "high_pass_filter_impl.h"
-#include "gain_control_impl.h"
-#include "level_estimator_impl.h"
-#include "module_common_types.h"
-#include "noise_suppression_impl.h"
-#include "processing_component.h"
-#include "splitting_filter.h"
-#include "voice_detection_impl.h"
-#ifndef NDEBUG
-#ifdef WEBRTC_ANDROID
-#include "external/webrtc/src/modules/audio_processing/main/source/debug.pb.h"
-#else
-#include "webrtc/audio_processing/debug.pb.h"
-#endif
-#endif /* NDEBUG */
-
-namespace webrtc {
-AudioProcessing* AudioProcessing::Create(int id) {
-  /*WEBRTC_TRACE(webrtc::kTraceModuleCall,
-             webrtc::kTraceAudioProcessing,
-             id,
-             "AudioProcessing::Create()");*/
-
-  AudioProcessingImpl* apm = new AudioProcessingImpl(id);
-  if (apm->Initialize() != kNoError) {
-    delete apm;
-    apm = NULL;
-  }
-
-  return apm;
-}
-
-void AudioProcessing::Destroy(AudioProcessing* apm) {
-  delete static_cast<AudioProcessingImpl*>(apm);
-}
-
-AudioProcessingImpl::AudioProcessingImpl(int id)
-    : id_(id),
-      echo_cancellation_(NULL),
-      echo_control_mobile_(NULL),
-      gain_control_(NULL),
-      high_pass_filter_(NULL),
-      level_estimator_(NULL),
-      noise_suppression_(NULL),
-      voice_detection_(NULL),
-#ifndef NDEBUG
-      debug_file_(FileWrapper::Create()),
-      event_msg_(new audioproc::Event()),
-#endif
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      render_audio_(NULL),
-      capture_audio_(NULL),
-      sample_rate_hz_(kSampleRate16kHz),
-      split_sample_rate_hz_(kSampleRate16kHz),
-      samples_per_channel_(sample_rate_hz_ / 100),
-      stream_delay_ms_(0),
-      was_stream_delay_set_(false),
-      num_reverse_channels_(1),
-      num_input_channels_(1),
-      num_output_channels_(1) {
-
-  echo_cancellation_ = new EchoCancellationImpl(this);
-  component_list_.push_back(echo_cancellation_);
-
-  echo_control_mobile_ = new EchoControlMobileImpl(this);
-  component_list_.push_back(echo_control_mobile_);
-
-  gain_control_ = new GainControlImpl(this);
-  component_list_.push_back(gain_control_);
-
-  high_pass_filter_ = new HighPassFilterImpl(this);
-  component_list_.push_back(high_pass_filter_);
-
-  level_estimator_ = new LevelEstimatorImpl(this);
-  component_list_.push_back(level_estimator_);
-
-  noise_suppression_ = new NoiseSuppressionImpl(this);
-  component_list_.push_back(noise_suppression_);
-
-  voice_detection_ = new VoiceDetectionImpl(this);
-  component_list_.push_back(voice_detection_);
-}
-
-AudioProcessingImpl::~AudioProcessingImpl() {
-  while (!component_list_.empty()) {
-    ProcessingComponent* component = component_list_.front();
-    component->Destroy();
-    delete component;
-    component_list_.pop_front();
-  }
-
-#ifndef NDEBUG
-  if (debug_file_->Open()) {
-    debug_file_->CloseFile();
-  }
-  delete debug_file_;
-  debug_file_ = NULL;
-
-  delete event_msg_;
-  event_msg_ = NULL;
-#endif
-
-  delete crit_;
-  crit_ = NULL;
-
-  if (render_audio_) {
-    delete render_audio_;
-    render_audio_ = NULL;
-  }
-
-  if (capture_audio_) {
-    delete capture_audio_;
-    capture_audio_ = NULL;
-  }
-}
-
-CriticalSectionWrapper* AudioProcessingImpl::crit() const {
-  return crit_;
-}
-
-int AudioProcessingImpl::split_sample_rate_hz() const {
-  return split_sample_rate_hz_;
-}
-
-int AudioProcessingImpl::Initialize() {
-  CriticalSectionScoped crit_scoped(*crit_);
-  return InitializeLocked();
-}
-
-int AudioProcessingImpl::InitializeLocked() {
-  if (render_audio_ != NULL) {
-    delete render_audio_;
-    render_audio_ = NULL;
-  }
-
-  if (capture_audio_ != NULL) {
-    delete capture_audio_;
-    capture_audio_ = NULL;
-  }
-
-  render_audio_ = new AudioBuffer(num_reverse_channels_,
-                                  samples_per_channel_);
-  capture_audio_ = new AudioBuffer(num_input_channels_,
-                                   samples_per_channel_);
-
-  was_stream_delay_set_ = false;
-
-  // Initialize all components.
-  std::list<ProcessingComponent*>::iterator it;
-  for (it = component_list_.begin(); it != component_list_.end(); it++) {
-    int err = (*it)->Initialize();
-    if (err != kNoError) {
-      return err;
-    }
-  }
-
-#ifndef NDEBUG
-  if (debug_file_->Open()) {
-    int err = WriteInitMessage();
-    if (err != kNoError) {
-      return err;
-    }
-  }
-#endif
-
-  return kNoError;
-}
-
-int AudioProcessingImpl::set_sample_rate_hz(int rate) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  if (rate != kSampleRate8kHz &&
-      rate != kSampleRate16kHz &&
-      rate != kSampleRate32kHz) {
-    return kBadParameterError;
-  }
-
-  sample_rate_hz_ = rate;
-  samples_per_channel_ = rate / 100;
-
-  if (sample_rate_hz_ == kSampleRate32kHz) {
-    split_sample_rate_hz_ = kSampleRate16kHz;
-  } else {
-    split_sample_rate_hz_ = sample_rate_hz_;
-  }
-
-  return InitializeLocked();
-}
-
-int AudioProcessingImpl::sample_rate_hz() const {
-  return sample_rate_hz_;
-}
-
-int AudioProcessingImpl::set_num_reverse_channels(int channels) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  // Only stereo supported currently.
-  if (channels > 2 || channels < 1) {
-    return kBadParameterError;
-  }
-
-  num_reverse_channels_ = channels;
-
-  return InitializeLocked();
-}
-
-int AudioProcessingImpl::num_reverse_channels() const {
-  return num_reverse_channels_;
-}
-
-int AudioProcessingImpl::set_num_channels(
-    int input_channels,
-    int output_channels) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  if (output_channels > input_channels) {
-    return kBadParameterError;
-  }
-
-  // Only stereo supported currently.
-  if (input_channels > 2 || input_channels < 1) {
-    return kBadParameterError;
-  }
-
-  if (output_channels > 2 || output_channels < 1) {
-    return kBadParameterError;
-  }
-
-  num_input_channels_ = input_channels;
-  num_output_channels_ = output_channels;
-
-  return InitializeLocked();
-}
-
-int AudioProcessingImpl::num_input_channels() const {
-  return num_input_channels_;
-}
-
-int AudioProcessingImpl::num_output_channels() const {
-  return num_output_channels_;
-}
-
-int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  int err = kNoError;
-
-  if (frame == NULL) {
-    return kNullPointerError;
-  }
-
-  if (frame->_frequencyInHz != sample_rate_hz_) {
-    return kBadSampleRateError;
-  }
-
-  if (frame->_audioChannel != num_input_channels_) {
-    return kBadNumberChannelsError;
-  }
-
-  if (frame->_payloadDataLengthInSamples != samples_per_channel_) {
-    return kBadDataLengthError;
-  }
-
-#ifndef NDEBUG
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::STREAM);
-    audioproc::Stream* msg = event_msg_->mutable_stream();
-    const size_t data_size = sizeof(WebRtc_Word16) *
-                             frame->_payloadDataLengthInSamples *
-                             frame->_audioChannel;
-    msg->set_input_data(frame->_payloadData, data_size);
-    msg->set_delay(stream_delay_ms_);
-    msg->set_drift(echo_cancellation_->stream_drift_samples());
-    msg->set_level(gain_control_->stream_analog_level());
-  }
-#endif
-
-  capture_audio_->DeinterleaveFrom(frame);
-
-  // TODO(ajm): experiment with mixing and AEC placement.
-  if (num_output_channels_ < num_input_channels_) {
-    capture_audio_->Mix(num_output_channels_);
-
-    frame->_audioChannel = num_output_channels_;
-  }
-
-  if (sample_rate_hz_ == kSampleRate32kHz) {
-    for (int i = 0; i < num_input_channels_; i++) {
-      // Split into a low and high band.
-      SplittingFilterAnalysis(capture_audio_->data(i),
-                              capture_audio_->low_pass_split_data(i),
-                              capture_audio_->high_pass_split_data(i),
-                              capture_audio_->analysis_filter_state1(i),
-                              capture_audio_->analysis_filter_state2(i));
-    }
-  }
-
-  err = high_pass_filter_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = gain_control_->AnalyzeCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = echo_cancellation_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  if (echo_control_mobile_->is_enabled() &&
-      noise_suppression_->is_enabled()) {
-    capture_audio_->CopyLowPassToReference();
-  }
-
-  err = noise_suppression_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = echo_control_mobile_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = voice_detection_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = gain_control_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  //err = level_estimator_->ProcessCaptureAudio(capture_audio_);
-  //if (err != kNoError) {
-  //  return err;
-  //}
-
-  if (sample_rate_hz_ == kSampleRate32kHz) {
-    for (int i = 0; i < num_output_channels_; i++) {
-      // Recombine low and high bands.
-      SplittingFilterSynthesis(capture_audio_->low_pass_split_data(i),
-                               capture_audio_->high_pass_split_data(i),
-                               capture_audio_->data(i),
-                               capture_audio_->synthesis_filter_state1(i),
-                               capture_audio_->synthesis_filter_state2(i));
-    }
-  }
-
-  capture_audio_->InterleaveTo(frame);
-
-#ifndef NDEBUG
-  if (debug_file_->Open()) {
-    audioproc::Stream* msg = event_msg_->mutable_stream();
-    const size_t data_size = sizeof(WebRtc_Word16) *
-                             frame->_payloadDataLengthInSamples *
-                             frame->_audioChannel;
-    msg->set_output_data(frame->_payloadData, data_size);
-    err = WriteMessageToDebugFile();
-    if (err != kNoError) {
-      return err;
-    }
-  }
-#endif
-
-  return kNoError;
-}
-
-int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  int err = kNoError;
-
-  if (frame == NULL) {
-    return kNullPointerError;
-  }
-
-  if (frame->_frequencyInHz != sample_rate_hz_) {
-    return kBadSampleRateError;
-  }
-
-  if (frame->_audioChannel != num_reverse_channels_) {
-    return kBadNumberChannelsError;
-  }
-
-  if (frame->_payloadDataLengthInSamples != samples_per_channel_) {
-    return kBadDataLengthError;
-  }
-
-#ifndef NDEBUG
-  if (debug_file_->Open()) {
-    event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
-    audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
-    const size_t data_size = sizeof(WebRtc_Word16) *
-                             frame->_payloadDataLengthInSamples *
-                             frame->_audioChannel;
-    msg->set_data(frame->_payloadData, data_size);
-    err = WriteMessageToDebugFile();
-    if (err != kNoError) {
-      return err;
-    }
-  }
-#endif
-
-  render_audio_->DeinterleaveFrom(frame);
-
-  // TODO(ajm): turn the splitting filter into a component?
-  if (sample_rate_hz_ == kSampleRate32kHz) {
-    for (int i = 0; i < num_reverse_channels_; i++) {
-      // Split into low and high band.
-      SplittingFilterAnalysis(render_audio_->data(i),
-                              render_audio_->low_pass_split_data(i),
-                              render_audio_->high_pass_split_data(i),
-                              render_audio_->analysis_filter_state1(i),
-                              render_audio_->analysis_filter_state2(i));
-    }
-  }
-
-  // TODO(ajm): warnings possible from components?
-  err = echo_cancellation_->ProcessRenderAudio(render_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = echo_control_mobile_->ProcessRenderAudio(render_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = gain_control_->ProcessRenderAudio(render_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  //err = level_estimator_->AnalyzeReverseStream(render_audio_);
-  //if (err != kNoError) {
-  //  return err;
-  //}
-
-  was_stream_delay_set_ = false;
-  return err;  // TODO(ajm): this is for returning warnings; necessary?
-}
-
-int AudioProcessingImpl::set_stream_delay_ms(int delay) {
-  was_stream_delay_set_ = true;
-  if (delay < 0) {
-    return kBadParameterError;
-  }
-
-  // TODO(ajm): the max is rather arbitrarily chosen; investigate.
-  if (delay > 500) {
-    stream_delay_ms_ = 500;
-    return kBadStreamParameterWarning;
-  }
-
-  stream_delay_ms_ = delay;
-  return kNoError;
-}
-
-int AudioProcessingImpl::stream_delay_ms() const {
-  return stream_delay_ms_;
-}
-
-bool AudioProcessingImpl::was_stream_delay_set() const {
-  return was_stream_delay_set_;
-}
-
-int AudioProcessingImpl::StartDebugRecording(
-    const char filename[AudioProcessing::kMaxFilenameSize]) {
-#ifndef NDEBUG
-  CriticalSectionScoped crit_scoped(*crit_);
-  assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize);
-
-  if (filename == NULL) {
-    return kNullPointerError;
-  }
-
-  // Stop any ongoing recording.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
-      return kFileError;
-    }
-  }
-
-  if (debug_file_->OpenFile(filename, false) == -1) {
-    debug_file_->CloseFile();
-    return kFileError;
-  }
-
-  int err = WriteInitMessage();
-  if (err != kNoError) {
-    return err;
-  }
-#endif
-
-  return kNoError;
-}
-
-int AudioProcessingImpl::StopDebugRecording() {
-#ifndef NDEBUG
-  CriticalSectionScoped crit_scoped(*crit_);
-  // We just return if recording hasn't started.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
-      return kFileError;
-    }
-  }
-#endif
-
-  return kNoError;
-}
-
-EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
-  return echo_cancellation_;
-}
-
-EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
-  return echo_control_mobile_;
-}
-
-GainControl* AudioProcessingImpl::gain_control() const {
-  return gain_control_;
-}
-
-HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
-  return high_pass_filter_;
-}
-
-LevelEstimator* AudioProcessingImpl::level_estimator() const {
-  return level_estimator_;
-}
-
-NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
-  return noise_suppression_;
-}
-
-VoiceDetection* AudioProcessingImpl::voice_detection() const {
-  return voice_detection_;
-}
-
-WebRtc_Word32 AudioProcessingImpl::Version(WebRtc_Word8* version,
-    WebRtc_UWord32& bytes_remaining, WebRtc_UWord32& position) const {
-  if (version == NULL) {
-    /*WEBRTC_TRACE(webrtc::kTraceError,
-               webrtc::kTraceAudioProcessing,
-               -1,
-               "Null version pointer");*/
-    return kNullPointerError;
-  }
-  memset(&version[position], 0, bytes_remaining);
-
-  char my_version[] = "AudioProcessing 1.0.0";
-  // Includes null termination.
-  WebRtc_UWord32 length = static_cast<WebRtc_UWord32>(strlen(my_version));
-  if (bytes_remaining < length) {
-    /*WEBRTC_TRACE(webrtc::kTraceError,
-               webrtc::kTraceAudioProcessing,
-               -1,
-               "Buffer of insufficient length");*/
-    return kBadParameterError;
-  }
-  memcpy(&version[position], my_version, length);
-  bytes_remaining -= length;
-  position += length;
-
-  std::list<ProcessingComponent*>::const_iterator it;
-  for (it = component_list_.begin(); it != component_list_.end(); it++) {
-    char component_version[256];
-    strcpy(component_version, "\n");
-    int err = (*it)->get_version(&component_version[1],
-                                 sizeof(component_version) - 1);
-    if (err != kNoError) {
-      return err;
-    }
-    if (strncmp(&component_version[1], "\0", 1) == 0) {
-      // Assume empty if first byte is NULL.
-      continue;
-    }
-
-    length = static_cast<WebRtc_UWord32>(strlen(component_version));
-    if (bytes_remaining < length) {
-      /*WEBRTC_TRACE(webrtc::kTraceError,
-                 webrtc::kTraceAudioProcessing,
-                 -1,
-                 "Buffer of insufficient length");*/
-      return kBadParameterError;
-    }
-    memcpy(&version[position], component_version, length);
-    bytes_remaining -= length;
-    position += length;
-  }
-
-  return kNoError;
-}
-
-WebRtc_Word32 AudioProcessingImpl::ChangeUniqueId(const WebRtc_Word32 id) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  /*WEBRTC_TRACE(webrtc::kTraceModuleCall,
-             webrtc::kTraceAudioProcessing,
-             id_,
-             "ChangeUniqueId(new id = %d)",
-             id);*/
-  id_ = id;
-
-  return kNoError;
-}
-
-#ifndef NDEBUG
-int AudioProcessingImpl::WriteMessageToDebugFile() {
-  int32_t size = event_msg_->ByteSize();
-  if (size <= 0) {
-    return kUnspecifiedError;
-  }
-#if defined(WEBRTC_BIG_ENDIAN)
-  // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
-  //            pretty safe in assuming little-endian.
-#endif
-
-  if (!event_msg_->SerializeToString(&event_str_)) {
-    return kUnspecifiedError;
-  }
-
-  // Write message preceded by its size.
-  if (!debug_file_->Write(&size, sizeof(int32_t))) {
-    return kFileError;
-  }
-  if (!debug_file_->Write(event_str_.data(), event_str_.length())) {
-    return kFileError;
-  }
-
-  event_msg_->Clear();
-
-  return 0;
-}
-
-int AudioProcessingImpl::WriteInitMessage() {
-  event_msg_->set_type(audioproc::Event::INIT);
-  audioproc::Init* msg = event_msg_->mutable_init();
-  msg->set_sample_rate(sample_rate_hz_);
-  msg->set_device_sample_rate(echo_cancellation_->device_sample_rate_hz());
-  msg->set_num_input_channels(num_input_channels_);
-  msg->set_num_output_channels(num_output_channels_);
-  msg->set_num_reverse_channels(num_reverse_channels_);
-
-  int err = WriteMessageToDebugFile();
-  if (err != kNoError) {
-    return err;
-  }
-
-  return kNoError;
-}
-#endif
-}  // namespace webrtc

src/modules/audio_processing/audio_processing_impl.h

@@ -1,117 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
-
-#include <list>
-#include <string>
-
-#include "audio_processing.h"
-
-namespace webrtc {
-namespace audioproc {
-class Event;
-}  // audioproc
-class AudioBuffer;
-class CriticalSectionWrapper;
-class EchoCancellationImpl;
-class EchoControlMobileImpl;
-class FileWrapper;
-class GainControlImpl;
-class HighPassFilterImpl;
-class LevelEstimatorImpl;
-class NoiseSuppressionImpl;
-class ProcessingComponent;
-class VoiceDetectionImpl;
-
-class AudioProcessingImpl : public AudioProcessing {
- public:
-  enum {
-    kSampleRate8kHz = 8000,
-    kSampleRate16kHz = 16000,
-    kSampleRate32kHz = 32000
-  };
-
-  explicit AudioProcessingImpl(int id);
-  virtual ~AudioProcessingImpl();
-
-  CriticalSectionWrapper* crit() const;
-
-  int split_sample_rate_hz() const;
-  bool was_stream_delay_set() const;
-
-  // AudioProcessing methods.
-  virtual int Initialize();
-  virtual int InitializeLocked();
-  virtual int set_sample_rate_hz(int rate);
-  virtual int sample_rate_hz() const;
-  virtual int set_num_channels(int input_channels, int output_channels);
-  virtual int num_input_channels() const;
-  virtual int num_output_channels() const;
-  virtual int set_num_reverse_channels(int channels);
-  virtual int num_reverse_channels() const;
-  virtual int ProcessStream(AudioFrame* frame);
-  virtual int AnalyzeReverseStream(AudioFrame* frame);
-  virtual int set_stream_delay_ms(int delay);
-  virtual int stream_delay_ms() const;
-  virtual int StartDebugRecording(const char filename[kMaxFilenameSize]);
-  virtual int StopDebugRecording();
-  virtual EchoCancellation* echo_cancellation() const;
-  virtual EchoControlMobile* echo_control_mobile() const;
-  virtual GainControl* gain_control() const;
-  virtual HighPassFilter* high_pass_filter() const;
-  virtual LevelEstimator* level_estimator() const;
-  virtual NoiseSuppression* noise_suppression() const;
-  virtual VoiceDetection* voice_detection() const;
-
-  // Module methods.
-  virtual WebRtc_Word32 Version(WebRtc_Word8* version,
-                              WebRtc_UWord32& remainingBufferInBytes,
-                              WebRtc_UWord32& position) const;
-  virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- private:
-  int WriteMessageToDebugFile();
-  int WriteInitMessage();
-
-  int id_;
-
-  EchoCancellationImpl* echo_cancellation_;
-  EchoControlMobileImpl* echo_control_mobile_;
-  GainControlImpl* gain_control_;
-  HighPassFilterImpl* high_pass_filter_;
-  LevelEstimatorImpl* level_estimator_;
-  NoiseSuppressionImpl* noise_suppression_;
-  VoiceDetectionImpl* voice_detection_;
-
-  std::list<ProcessingComponent*> component_list_;
-
-  FileWrapper* debug_file_;
-  audioproc::Event* event_msg_; // Protobuf message.
-  std::string event_str_; // Memory for protobuf serialization.
-  CriticalSectionWrapper* crit_;
-
-  AudioBuffer* render_audio_;
-  AudioBuffer* capture_audio_;
-
-  int sample_rate_hz_;
-  int split_sample_rate_hz_;
-  int samples_per_channel_;
-  int stream_delay_ms_;
-  bool was_stream_delay_set_;
-
-  int num_reverse_channels_;
-  int num_input_channels_;
-  int num_output_channels_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_

src/modules/audio_processing/debug.proto

@@ -1,37 +0,0 @@
-syntax = "proto2";
-option optimize_for = LITE_RUNTIME;
-package webrtc.audioproc;
-
-message Init {
-  optional int32 sample_rate = 1;
-  optional int32 device_sample_rate = 2;
-  optional int32 num_input_channels = 3;
-  optional int32 num_output_channels = 4;
-  optional int32 num_reverse_channels = 5;
-}
-
-message ReverseStream {
-  optional bytes data = 1;
-}
-
-message Stream {
-  optional bytes input_data = 1;
-  optional bytes output_data = 2;
-  optional int32 delay = 3;
-  optional sint32 drift = 4;
-  optional int32 level = 5;
-}
-
-message Event {
-  enum Type {
-    INIT = 0;
-    REVERSE_STREAM = 1;
-    STREAM = 2;
-  }
-
-  required Type type = 1;
-
-  optional Init init = 2;
-  optional ReverseStream reverse_stream = 3;
-  optional Stream stream = 4;
-}

src/modules/audio_processing/echo_cancellation_impl.cc

@@ -1,383 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "echo_cancellation_impl.h"
-
-#include <cassert>
-#include <string.h>
-
-#include "critical_section_wrapper.h"
-#include "echo_cancellation.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-
-typedef void Handle;
-
-namespace {
-WebRtc_Word16 MapSetting(EchoCancellation::SuppressionLevel level) {
-  switch (level) {
-    case EchoCancellation::kLowSuppression:
-      return kAecNlpConservative;
-    case EchoCancellation::kModerateSuppression:
-      return kAecNlpModerate;
-    case EchoCancellation::kHighSuppression:
-      return kAecNlpAggressive;
-    default:
-      return -1;
-  }
-}
-
-int MapError(int err) {
-  switch (err) {
-    case AEC_UNSUPPORTED_FUNCTION_ERROR:
-      return AudioProcessing::kUnsupportedFunctionError;
-      break;
-    case AEC_BAD_PARAMETER_ERROR:
-      return AudioProcessing::kBadParameterError;
-      break;
-    case AEC_BAD_PARAMETER_WARNING:
-      return AudioProcessing::kBadStreamParameterWarning;
-      break;
-    default:
-      // AEC_UNSPECIFIED_ERROR
-      // AEC_UNINITIALIZED_ERROR
-      // AEC_NULL_POINTER_ERROR
-      return AudioProcessing::kUnspecifiedError;
-  }
-}
-}  // namespace
-
-EchoCancellationImpl::EchoCancellationImpl(const AudioProcessingImpl* apm)
-  : ProcessingComponent(apm),
-    apm_(apm),
-    drift_compensation_enabled_(false),
-    metrics_enabled_(false),
-    suppression_level_(kModerateSuppression),
-    device_sample_rate_hz_(48000),
-    stream_drift_samples_(0),
-    was_stream_drift_set_(false),
-    stream_has_echo_(false),
-    delay_logging_enabled_(false) {}
-
-EchoCancellationImpl::~EchoCancellationImpl() {}
-
-int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  assert(audio->samples_per_split_channel() <= 160);
-  assert(audio->num_channels() == apm_->num_reverse_channels());
-
-  int err = apm_->kNoError;
-
-  // The ordering convention must be followed to pass to the correct AEC.
-  size_t handle_index = 0;
-  for (int i = 0; i < apm_->num_output_channels(); i++) {
-    for (int j = 0; j < audio->num_channels(); j++) {
-      Handle* my_handle = static_cast<Handle*>(handle(handle_index));
-      err = WebRtcAec_BufferFarend(
-          my_handle,
-          audio->low_pass_split_data(j),
-          static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
-
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);  // TODO(ajm): warning possible?
-      }
-
-      handle_index++;
-    }
-  }
-
-  return apm_->kNoError;
-}
-
-int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  if (!apm_->was_stream_delay_set()) {
-    return apm_->kStreamParameterNotSetError;
-  }
-
-  if (drift_compensation_enabled_ && !was_stream_drift_set_) {
-    return apm_->kStreamParameterNotSetError;
-  }
-
-  assert(audio->samples_per_split_channel() <= 160);
-  assert(audio->num_channels() == apm_->num_output_channels());
-
-  int err = apm_->kNoError;
-
-  // The ordering convention must be followed to pass to the correct AEC.
-  size_t handle_index = 0;
-  stream_has_echo_ = false;
-  for (int i = 0; i < audio->num_channels(); i++) {
-    for (int j = 0; j < apm_->num_reverse_channels(); j++) {
-      Handle* my_handle = handle(handle_index);
-      err = WebRtcAec_Process(
-          my_handle,
-          audio->low_pass_split_data(i),
-          audio->high_pass_split_data(i),
-          audio->low_pass_split_data(i),
-          audio->high_pass_split_data(i),
-          static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
-          apm_->stream_delay_ms(),
-          stream_drift_samples_);
-
-      if (err != apm_->kNoError) {
-        err = GetHandleError(my_handle);
-        // TODO(ajm): Figure out how to return warnings properly.
-        if (err != apm_->kBadStreamParameterWarning) {
-          return err;
-        }
-      }
-
-      WebRtc_Word16 status = 0;
-      err = WebRtcAec_get_echo_status(my_handle, &status);
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);
-      }
-
-      if (status == 1) {
-        stream_has_echo_ = true;
-      }
-
-      handle_index++;
-    }
-  }
-
-  was_stream_drift_set_ = false;
-  return apm_->kNoError;
-}
-
-int EchoCancellationImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  // Ensure AEC and AECM are not both enabled.
-  if (enable && apm_->echo_control_mobile()->is_enabled()) {
-    return apm_->kBadParameterError;
-  }
-
-  return EnableComponent(enable);
-}
-
-bool EchoCancellationImpl::is_enabled() const {
-  return is_component_enabled();
-}
-
-int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (MapSetting(level) == -1) {
-    return apm_->kBadParameterError;
-  }
-
-  suppression_level_ = level;
-  return Configure();
-}
-
-EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level()
-    const {
-  return suppression_level_;
-}
-
-int EchoCancellationImpl::enable_drift_compensation(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  drift_compensation_enabled_ = enable;
-  return Configure();
-}
-
-bool EchoCancellationImpl::is_drift_compensation_enabled() const {
-  return drift_compensation_enabled_;
-}
-
-int EchoCancellationImpl::set_device_sample_rate_hz(int rate) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (rate < 8000 || rate > 96000) {
-    return apm_->kBadParameterError;
-  }
-
-  device_sample_rate_hz_ = rate;
-  return Initialize();
-}
-
-int EchoCancellationImpl::device_sample_rate_hz() const {
-  return device_sample_rate_hz_;
-}
-
-int EchoCancellationImpl::set_stream_drift_samples(int drift) {
-  was_stream_drift_set_ = true;
-  stream_drift_samples_ = drift;
-  return apm_->kNoError;
-}
-
-int EchoCancellationImpl::stream_drift_samples() const {
-  return stream_drift_samples_;
-}
-
-int EchoCancellationImpl::enable_metrics(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  metrics_enabled_ = enable;
-  return Configure();
-}
-
-bool EchoCancellationImpl::are_metrics_enabled() const {
-  return metrics_enabled_;
-}
-
-// TODO(ajm): we currently just use the metrics from the first AEC. Think more
-//            aboue the best way to extend this to multi-channel.
-int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (metrics == NULL) {
-    return apm_->kNullPointerError;
-  }
-
-  if (!is_component_enabled() || !metrics_enabled_) {
-    return apm_->kNotEnabledError;
-  }
-
-  AecMetrics my_metrics;
-  memset(&my_metrics, 0, sizeof(my_metrics));
-  memset(metrics, 0, sizeof(Metrics));
-
-  Handle* my_handle = static_cast<Handle*>(handle(0));
-  int err = WebRtcAec_GetMetrics(my_handle, &my_metrics);
-  if (err != apm_->kNoError) {
-    return GetHandleError(my_handle);
-  }
-
-  metrics->residual_echo_return_loss.instant = my_metrics.rerl.instant;
-  metrics->residual_echo_return_loss.average = my_metrics.rerl.average;
-  metrics->residual_echo_return_loss.maximum = my_metrics.rerl.max;
-  metrics->residual_echo_return_loss.minimum = my_metrics.rerl.min;
-
-  metrics->echo_return_loss.instant = my_metrics.erl.instant;
-  metrics->echo_return_loss.average = my_metrics.erl.average;
-  metrics->echo_return_loss.maximum = my_metrics.erl.max;
-  metrics->echo_return_loss.minimum = my_metrics.erl.min;
-
-  metrics->echo_return_loss_enhancement.instant = my_metrics.erle.instant;
-  metrics->echo_return_loss_enhancement.average = my_metrics.erle.average;
-  metrics->echo_return_loss_enhancement.maximum = my_metrics.erle.max;
-  metrics->echo_return_loss_enhancement.minimum = my_metrics.erle.min;
-
-  metrics->a_nlp.instant = my_metrics.aNlp.instant;
-  metrics->a_nlp.average = my_metrics.aNlp.average;
-  metrics->a_nlp.maximum = my_metrics.aNlp.max;
-  metrics->a_nlp.minimum = my_metrics.aNlp.min;
-
-  return apm_->kNoError;
-}
-
-bool EchoCancellationImpl::stream_has_echo() const {
-  return stream_has_echo_;
-}
-
-int EchoCancellationImpl::enable_delay_logging(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  delay_logging_enabled_ = enable;
-  return Configure();
-}
-
-bool EchoCancellationImpl::is_delay_logging_enabled() const {
-  return delay_logging_enabled_;
-}
-
-// TODO(bjornv): How should we handle the multi-channel case?
-int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (median == NULL) {
-    return apm_->kNullPointerError;
-  }
-  if (std == NULL) {
-    return apm_->kNullPointerError;
-  }
-
-  if (!is_component_enabled() || !delay_logging_enabled_) {
-    return apm_->kNotEnabledError;
-  }
-
-  Handle* my_handle = static_cast<Handle*>(handle(0));
-  if (WebRtcAec_GetDelayMetrics(my_handle, median, std) !=
-      apm_->kNoError) {
-    return GetHandleError(my_handle);
-  }
-
-  return apm_->kNoError;
-}
-
-int EchoCancellationImpl::Initialize() {
-  int err = ProcessingComponent::Initialize();
-  if (err != apm_->kNoError || !is_component_enabled()) {
-    return err;
-  }
-
-  was_stream_drift_set_ = false;
-
-  return apm_->kNoError;
-}
-
-int EchoCancellationImpl::get_version(char* version,
-                                      int version_len_bytes) const {
-  if (WebRtcAec_get_version(version, version_len_bytes) != 0) {
-      return apm_->kBadParameterError;
-  }
-
-  return apm_->kNoError;
-}
-
-void* EchoCancellationImpl::CreateHandle() const {
-  Handle* handle = NULL;
-  if (WebRtcAec_Create(&handle) != apm_->kNoError) {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
-}
-
-int EchoCancellationImpl::DestroyHandle(void* handle) const {
-  assert(handle != NULL);
-  return WebRtcAec_Free(static_cast<Handle*>(handle));
-}
-
-int EchoCancellationImpl::InitializeHandle(void* handle) const {
-  assert(handle != NULL);
-  return WebRtcAec_Init(static_cast<Handle*>(handle),
-                       apm_->sample_rate_hz(),
-                       device_sample_rate_hz_);
-}
-
-int EchoCancellationImpl::ConfigureHandle(void* handle) const {
-  assert(handle != NULL);
-  AecConfig config;
-  config.metricsMode = metrics_enabled_;
-  config.nlpMode = MapSetting(suppression_level_);
-  config.skewMode = drift_compensation_enabled_;
-  config.delay_logging = delay_logging_enabled_;
-
-  return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
-}
-
-int EchoCancellationImpl::num_handles_required() const {
-  return apm_->num_output_channels() *
-         apm_->num_reverse_channels();
-}
-
-int EchoCancellationImpl::GetHandleError(void* handle) const {
-  assert(handle != NULL);
-  return MapError(WebRtcAec_get_error_code(static_cast<Handle*>(handle)));
-}
-}  // namespace webrtc