From 2e405277c6c49eb717270e8cf19c3b79e0aeb0f4 Mon Sep 17 00:00:00 2001
From: Andre Natal <anatal@gmail.com>
Date: Tue, 28 Aug 2018 19:09:40 -0700
Subject: [PATCH] Complete Implementation

---
 .gitignore                                    |   97 +
 app/.gitignore                                |    1 +
 app/build.gradle                              |   34 +
 app/proguard-rules.pro                        |   21 +
 .../speechmodule/ExampleInstrumentedTest.java |   26 +
 app/src/main/AndroidManifest.xml              |   26 +
 .../com/mozilla/speechapp/MainActivity.java   |  128 ++
 .../drawable-v24/ic_launcher_foreground.xml   |   34 +
 .../res/drawable/ic_launcher_background.xml   |  170 ++
 app/src/main/res/layout/activity_main.xml     |   52 +
 .../res/mipmap-anydpi-v26/ic_launcher.xml     |    5 +
 .../mipmap-anydpi-v26/ic_launcher_round.xml   |    5 +
 app/src/main/res/mipmap-hdpi/ic_launcher.png  |  Bin 0 -> 3056 bytes
 .../res/mipmap-hdpi/ic_launcher_round.png     |  Bin 0 -> 5024 bytes
 app/src/main/res/mipmap-mdpi/ic_launcher.png  |  Bin 0 -> 2096 bytes
 .../res/mipmap-mdpi/ic_launcher_round.png     |  Bin 0 -> 2858 bytes
 app/src/main/res/mipmap-xhdpi/ic_launcher.png |  Bin 0 -> 4569 bytes
 .../res/mipmap-xhdpi/ic_launcher_round.png    |  Bin 0 -> 7098 bytes
 .../main/res/mipmap-xxhdpi/ic_launcher.png    |  Bin 0 -> 6464 bytes
 .../res/mipmap-xxhdpi/ic_launcher_round.png   |  Bin 0 -> 10676 bytes
 .../main/res/mipmap-xxxhdpi/ic_launcher.png   |  Bin 0 -> 9250 bytes
 .../res/mipmap-xxxhdpi/ic_launcher_round.png  |  Bin 0 -> 15523 bytes
 app/src/main/res/values/colors.xml            |    6 +
 app/src/main/res/values/strings.xml           |    3 +
 app/src/main/res/values/styles.xml            |   11 +
 .../mozilla/speechmodule/ExampleUnitTest.java |   17 +
 build.gradle                                  |   28 +
 gradle.properties                             |   29 +
 gradle/wrapper/gradle-wrapper.properties      |    6 +
 gradlew                                       |  172 ++
 gradlew.bat                                   |   84 +
 mozillaspeechlibrary/.gitignore               |    1 +
 mozillaspeechlibrary/build.gradle             |   49 +
 mozillaspeechlibrary/gradle.properties        |    3 +
 mozillaspeechlibrary/maven-push.gradle        |  112 ++
 mozillaspeechlibrary/proguard-rules.pro       |   21 +
 .../ExampleInstrumentedTest.java              |   26 +
 .../src/main/AndroidManifest.xml              |    2 +
 mozillaspeechlibrary/src/main/cpp/Android.mk  |   37 +
 .../src/main/cpp/cpu_features_wrapper.h       |   51 +
 .../src/main/cpp/signal_processing_library.h  | 1686 ++++++++++++++++
 mozillaspeechlibrary/src/main/cpp/spl_inl.h   |  159 ++
 .../src/main/cpp/spl_inl_armv7.h              |  137 ++
 mozillaspeechlibrary/src/main/cpp/typedefs.h  |  145 ++
 .../cpp/webrtc/common_audio/audio_util.cc     |   41 +
 .../webrtc/common_audio/include/audio_util.h  |   47 +
 .../webrtc/common_audio/include/typedefs.h    |  112 ++
 .../common_audio/signal_processing/Android.mk |  126 ++
 .../signal_processing/complex_bit_reverse.c   |  108 +
 .../signal_processing/complex_fft.c           |  307 +++
 .../signal_processing/complex_fft_tables.h    |  148 ++
 .../signal_processing/cross_correlation.c     |   31 +
 .../signal_processing/division_operations.c   |  143 ++
 .../signal_processing/downsample_fast.c       |   48 +
 .../common_audio/signal_processing/energy.c   |   36 +
 .../signal_processing/get_scaling_square.c    |   44 +
 .../signal_processing/include/real_fft.h      |  128 ++
 .../include/signal_processing_library.h       | 1779 +++++++++++++++++
 .../signal_processing/include/spl_inl.h       |  177 ++
 .../signal_processing/include/spl_inl_armv7.h |  165 ++
 .../signal_processing/include/spl_inl_mips.h  |  281 +++
 .../signal_processing/min_max_operations.c    |  243 +++
 .../common_audio/signal_processing/real_fft.c |  126 ++
 .../common_audio/signal_processing/resample.c |  505 +++++
 .../signal_processing/resample_48khz.c        |  186 ++
 .../signal_processing/resample_by_2.c         |  183 ++
 .../resample_by_2_internal.c                  |  679 +++++++
 .../resample_by_2_internal.h                  |   47 +
 .../signal_processing/resample_fractional.c   |  242 +++
 .../common_audio/signal_processing/spl_init.c |  157 ++
 .../common_audio/signal_processing/spl_sqrt.c |  184 ++
 .../signal_processing/spl_version.c           |   25 +
 .../sqrt_of_one_minus_x_squared.c             |   35 +
 .../vector_scaling_operations.c               |  169 ++
 .../cpp/webrtc/common_audio/vad/Android.mk    |   51 +
 .../common_audio/vad/include/webrtc_vad.h     |   91 +
 .../cpp/webrtc/common_audio/vad/vad_core.c    |  682 +++++++
 .../cpp/webrtc/common_audio/vad/vad_core.h    |  115 ++
 .../webrtc/common_audio/vad/vad_filterbank.c  |  334 ++++
 .../webrtc/common_audio/vad/vad_filterbank.h  |   44 +
 .../cpp/webrtc/common_audio/vad/vad_gmm.c     |   83 +
 .../cpp/webrtc/common_audio/vad/vad_gmm.h     |   39 +
 .../main/cpp/webrtc/common_audio/vad/vad_sp.c |  179 ++
 .../main/cpp/webrtc/common_audio/vad/vad_sp.h |   56 +
 .../cpp/webrtc/common_audio/vad/webrtc_vad.c  |  135 ++
 .../src/main/cpp/webrtc_jni.c                 |   43 +
 .../src/main/cpp/webrtc_vad.h                 |  159 ++
 .../axet/audiolibrary/encoders/Encoder.java   |    9 +
 .../audiolibrary/encoders/EncoderInfo.java    |   14 +
 .../axet/audiolibrary/encoders/Factory.java   |   46 +
 .../audiolibrary/encoders/FormatOPUS.java     |  173 ++
 .../audiolibrary/encoders/FormatOPUS_OGG.java |   85 +
 .../axet/audiolibrary/encoders/Natives.java   |   94 +
 .../axet/audiolibrary/encoders/Resample.java  |  104 +
 .../axet/audiolibrary/encoders/Sound.java     |   74 +
 .../ISpeechRecognitionListener.java           |    5 +
 .../speechlibrary/MozillaSpeechService.java   |   80 +
 .../com/mozilla/speechlibrary/Networking.java |   94 +
 .../com/mozilla/speechlibrary/STTResult.java  |   12 +
 .../speechlibrary/SpeechRecognition.java      |  140 ++
 .../java/com/mozilla/speechlibrary/Vad.java   |   13 +
 .../src/main/res/values/strings.xml           |    3 +
 .../speechlibrary/ExampleUnitTest.java        |   17 +
 settings.gradle                               |    1 +
 104 files changed, 12861 insertions(+)
 create mode 100644 .gitignore
 create mode 100644 app/.gitignore
 create mode 100644 app/build.gradle
 create mode 100644 app/proguard-rules.pro
 create mode 100644 app/src/androidTest/java/com/mozilla/speechmodule/ExampleInstrumentedTest.java
 create mode 100644 app/src/main/AndroidManifest.xml
 create mode 100644 app/src/main/java/com/mozilla/speechapp/MainActivity.java
 create mode 100644 app/src/main/res/drawable-v24/ic_launcher_foreground.xml
 create mode 100644 app/src/main/res/drawable/ic_launcher_background.xml
 create mode 100644 app/src/main/res/layout/activity_main.xml
 create mode 100644 app/src/main/res/mipmap-anydpi-v26/ic_launcher.xml
 create mode 100644 app/src/main/res/mipmap-anydpi-v26/ic_launcher_round.xml
 create mode 100644 app/src/main/res/mipmap-hdpi/ic_launcher.png
 create mode 100644 app/src/main/res/mipmap-hdpi/ic_launcher_round.png
 create mode 100644 app/src/main/res/mipmap-mdpi/ic_launcher.png
 create mode 100644 app/src/main/res/mipmap-mdpi/ic_launcher_round.png
 create mode 100644 app/src/main/res/mipmap-xhdpi/ic_launcher.png
 create mode 100644 app/src/main/res/mipmap-xhdpi/ic_launcher_round.png
 create mode 100644 app/src/main/res/mipmap-xxhdpi/ic_launcher.png
 create mode 100644 app/src/main/res/mipmap-xxhdpi/ic_launcher_round.png
 create mode 100644 app/src/main/res/mipmap-xxxhdpi/ic_launcher.png
 create mode 100644 app/src/main/res/mipmap-xxxhdpi/ic_launcher_round.png
 create mode 100644 app/src/main/res/values/colors.xml
 create mode 100644 app/src/main/res/values/strings.xml
 create mode 100644 app/src/main/res/values/styles.xml
 create mode 100644 app/src/test/java/com/mozilla/speechmodule/ExampleUnitTest.java
 create mode 100644 build.gradle
 create mode 100644 gradle.properties
 create mode 100644 gradle/wrapper/gradle-wrapper.properties
 create mode 100755 gradlew
 create mode 100644 gradlew.bat
 create mode 100644 mozillaspeechlibrary/.gitignore
 create mode 100644 mozillaspeechlibrary/build.gradle
 create mode 100644 mozillaspeechlibrary/gradle.properties
 create mode 100644 mozillaspeechlibrary/maven-push.gradle
 create mode 100644 mozillaspeechlibrary/proguard-rules.pro
 create mode 100644 mozillaspeechlibrary/src/androidTest/java/com/mozilla/speechlibrary/ExampleInstrumentedTest.java
 create mode 100644 mozillaspeechlibrary/src/main/AndroidManifest.xml
 create mode 100644 mozillaspeechlibrary/src/main/cpp/Android.mk
 create mode 100644 mozillaspeechlibrary/src/main/cpp/cpu_features_wrapper.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/signal_processing_library.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/spl_inl.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/spl_inl_armv7.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/typedefs.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/audio_util.cc
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/include/audio_util.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/include/typedefs.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/Android.mk
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_bit_reverse.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_fft.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_fft_tables.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/cross_correlation.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/division_operations.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/downsample_fast.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/energy.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/get_scaling_square.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/real_fft.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/signal_processing_library.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl_mips.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/min_max_operations.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/real_fft.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_48khz.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2_internal.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2_internal.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_fractional.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_init.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_sqrt.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_version.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/vector_scaling_operations.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/Android.mk
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/include/webrtc_vad.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_core.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_core.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_filterbank.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_filterbank.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_gmm.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_gmm.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_sp.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_sp.h
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/webrtc_vad.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc_jni.c
 create mode 100644 mozillaspeechlibrary/src/main/cpp/webrtc_vad.h
 create mode 100644 mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Encoder.java
 create mode 100644 mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/EncoderInfo.java
 create mode 100644 mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Factory.java
 create mode 100755 mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/FormatOPUS.java
 create mode 100755 mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/FormatOPUS_OGG.java
 create mode 100644 mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Natives.java
 create mode 100644 mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Resample.java
 create mode 100644 mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Sound.java
 create mode 100644 mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/ISpeechRecognitionListener.java
 create mode 100644 mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/MozillaSpeechService.java
 create mode 100644 mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/Networking.java
 create mode 100644 mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/STTResult.java
 create mode 100644 mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/SpeechRecognition.java
 create mode 100644 mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/Vad.java
 create mode 100644 mozillaspeechlibrary/src/main/res/values/strings.xml
 create mode 100644 mozillaspeechlibrary/src/test/java/com/mozilla/speechlibrary/ExampleUnitTest.java
 create mode 100644 settings.gradle

diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000..25cbd61
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1,97 @@
+*.iml
+.gradle
+/local.properties
+/.idea/libraries
+/.idea/modules.xml
+/.idea/workspace.xml
+.DS_Store
+/build
+/captures
+.externalNativeBuild
+
+# [Android] ========================
+# Built application files
+*.apk
+*.ap_
+
+# Files for the Dalvik VM
+*.dex
+
+# Java class files
+*.class
+
+# Generated files
+bin/
+gen/
+
+# Gradle files
+.gradle/
+build/
+
+# Local configuration file (sdk path, etc)
+local.properties
+
+# Proguard folder generated by Eclipse
+proguard/
+
+# Log Files
+*.log
+
+
+## Directory-based project format:
+.idea/
+
+## File-based project format:
+*.ipr
+*.iws
+
+## Plugin-specific files:
+
+# IntelliJ
+out/
+
+# mpeltonen/sbt-idea plugin
+.idea_modules/
+
+# JIRA plugin
+atlassian-ide-plugin.xml
+
+# Crashlytics plugin (for Android Studio and IntelliJ)
+com_crashlytics_export_strings.xml
+
+
+# [Maven] ========================
+target/
+pom.xml.tag
+pom.xml.releaseBackup
+pom.xml.versionsBackup
+pom.xml.next
+release.properties
+
+
+# [Gradle-Android] ========================
+
+# Ignore Gradle GUI config
+gradle-app.setting
+
+# Gradle Signing
+signing.properties
+trestle.keystore
+
+# Mobile Tools for Java (J2ME)
+.mtj.tmp/
+
+# Package Files #
+*.jar
+*.war
+*.ear
+
+# virtual machine crash logs, see http://www.java.com/en/download/help/error_hotspot.xml
+hs_err_pid*
+
+# Misc
+/.idea/workspace.xml
+.DS_Store
+/captures
+**/*.iml
+*.class
diff --git a/app/.gitignore b/app/.gitignore
new file mode 100644
index 0000000..796b96d
--- /dev/null
+++ b/app/.gitignore
@@ -0,0 +1 @@
+/build
diff --git a/app/build.gradle b/app/build.gradle
new file mode 100644
index 0000000..cbe5ca8
--- /dev/null
+++ b/app/build.gradle
@@ -0,0 +1,34 @@
+apply plugin: 'com.android.application'
+
+android {
+    compileSdkVersion 27
+    defaultConfig {
+        applicationId "com.mozilla.speechmodule"
+        minSdkVersion 24
+        targetSdkVersion 27
+        versionCode 1
+        versionName "1.0"
+        testInstrumentationRunner "android.support.test.runner.AndroidJUnitRunner"
+    }
+    buildTypes {
+        release {
+            minifyEnabled false
+            proguardFiles getDefaultProguardFile('proguard-android.txt'), 'proguard-rules.pro'
+        }
+    }
+    compileOptions {
+        sourceCompatibility JavaVersion.VERSION_1_8
+        targetCompatibility JavaVersion.VERSION_1_8
+    }
+}
+
+dependencies {
+    implementation project(':mozillaspeechlibrary')
+    implementation fileTree(include: ['*.jar'], dir: 'libs')
+    implementation 'com.android.support:appcompat-v7:27.1.1'
+    implementation 'com.android.support.constraint:constraint-layout:1.1.2'
+    implementation 'com.jjoe64:graphview:4.2.2'
+    testImplementation 'junit:junit:4.12'
+    androidTestImplementation 'com.android.support.test:runner:1.0.2'
+    androidTestImplementation 'com.android.support.test.espresso:espresso-core:3.0.2'
+}
diff --git a/app/proguard-rules.pro b/app/proguard-rules.pro
new file mode 100644
index 0000000..f1b4245
--- /dev/null
+++ b/app/proguard-rules.pro
@@ -0,0 +1,21 @@
+# Add project specific ProGuard rules here.
+# You can control the set of applied configuration files using the
+# proguardFiles setting in build.gradle.
+#
+# For more details, see
+#   http://developer.android.com/guide/developing/tools/proguard.html
+
+# If your project uses WebView with JS, uncomment the following
+# and specify the fully qualified class name to the JavaScript interface
+# class:
+#-keepclassmembers class fqcn.of.javascript.interface.for.webview {
+#   public *;
+#}
+
+# Uncomment this to preserve the line number information for
+# debugging stack traces.
+#-keepattributes SourceFile,LineNumberTable
+
+# If you keep the line number information, uncomment this to
+# hide the original source file name.
+#-renamesourcefileattribute SourceFile
diff --git a/app/src/androidTest/java/com/mozilla/speechmodule/ExampleInstrumentedTest.java b/app/src/androidTest/java/com/mozilla/speechmodule/ExampleInstrumentedTest.java
new file mode 100644
index 0000000..4cee819
--- /dev/null
+++ b/app/src/androidTest/java/com/mozilla/speechmodule/ExampleInstrumentedTest.java
@@ -0,0 +1,26 @@
+package com.mozilla.speechmodule;
+
+import android.content.Context;
+import android.support.test.InstrumentationRegistry;
+import android.support.test.runner.AndroidJUnit4;
+
+import org.junit.Test;
+import org.junit.runner.RunWith;
+
+import static org.junit.Assert.*;
+
+/**
+ * Instrumented test, which will execute on an Android device.
+ *
+ * @see <a href="http://d.android.com/tools/testing">Testing documentation</a>
+ */
+@RunWith(AndroidJUnit4.class)
+public class ExampleInstrumentedTest {
+    @Test
+    public void useAppContext() {
+        // Context of the app under test.
+        Context appContext = InstrumentationRegistry.getTargetContext();
+
+        assertEquals("com.mozilla.speechmodule", appContext.getPackageName());
+    }
+}
diff --git a/app/src/main/AndroidManifest.xml b/app/src/main/AndroidManifest.xml
new file mode 100644
index 0000000..7b2e730
--- /dev/null
+++ b/app/src/main/AndroidManifest.xml
@@ -0,0 +1,26 @@
+<?xml version="1.0" encoding="utf-8"?>
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+    package="com.mozilla.speechmodule">
+
+    <uses-permission android:name="android.permission.RECORD_AUDIO"></uses-permission>
+    <uses-permission android:name="android.permission.INTERNET"></uses-permission>
+
+    <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE" />
+    <uses-permission android:name="android.permission.READ_EXTERNAL_STORAGE" />
+    <application
+        android:allowBackup="true"
+        android:icon="@mipmap/ic_launcher"
+        android:label="@string/app_name"
+        android:roundIcon="@mipmap/ic_launcher_round"
+        android:supportsRtl="true"
+        android:theme="@style/AppTheme">
+        <activity android:name="com.mozilla.speechapp.MainActivity">
+            <intent-filter>
+                <action android:name="android.intent.action.MAIN" />
+
+                <category android:name="android.intent.category.LAUNCHER" />
+            </intent-filter>
+        </activity>
+    </application>
+
+</manifest>
\ No newline at end of file
diff --git a/app/src/main/java/com/mozilla/speechapp/MainActivity.java b/app/src/main/java/com/mozilla/speechapp/MainActivity.java
new file mode 100644
index 0000000..d971073
--- /dev/null
+++ b/app/src/main/java/com/mozilla/speechapp/MainActivity.java
@@ -0,0 +1,128 @@
+package com.mozilla.speechapp;
+
+import android.Manifest;
+import android.app.Activity;
+import android.content.pm.PackageManager;
+import android.support.annotation.NonNull;
+import android.support.v4.app.ActivityCompat;
+import android.support.v7.app.AppCompatActivity;
+import android.os.Bundle;
+import android.util.Log;
+import android.view.View;
+import android.view.WindowManager;
+import android.widget.Button;
+import android.widget.EditText;
+
+import com.jjoe64.graphview.GraphView;
+import com.jjoe64.graphview.series.DataPoint;
+import com.jjoe64.graphview.series.LineGraphSeries;
+import com.mozilla.speechlibrary.ISpeechRecognitionListener;
+import com.mozilla.speechlibrary.MozillaSpeechService;
+import com.mozilla.speechlibrary.STTResult;
+import com.mozilla.speechmodule.R;
+
+import static android.support.constraint.Constraints.TAG;
+
+public class MainActivity extends AppCompatActivity implements ISpeechRecognitionListener {
+
+    private Button mButtonStart, mButtonCancel;
+    private MozillaSpeechService mMozillaSpeechService;
+    private GraphView mGraph;
+    private long mDtstart;
+    private LineGraphSeries<DataPoint> mSeries1;
+    private EditText mPlain_text_input;
+
+    @Override
+    protected void onCreate(Bundle savedInstanceState) {
+        super.onCreate(savedInstanceState);
+        setContentView(R.layout.activity_main);
+        mMozillaSpeechService = MozillaSpeechService.getInstance();
+        mMozillaSpeechService.addListener(this);
+        initialize();
+    }
+
+    private void initialize() {
+
+        if (ActivityCompat.checkSelfPermission(this, Manifest.permission.RECORD_AUDIO)
+                != PackageManager.PERMISSION_GRANTED) {
+
+            ActivityCompat.requestPermissions(this, new String[]{Manifest.permission.RECORD_AUDIO},
+                    123);
+        }
+
+        if (ActivityCompat.checkSelfPermission(this, Manifest.permission.WRITE_EXTERNAL_STORAGE)
+                != PackageManager.PERMISSION_GRANTED) {
+
+            ActivityCompat.requestPermissions(this, new String[]{Manifest.permission.WRITE_EXTERNAL_STORAGE},
+                    124);
+        }
+        mButtonStart = findViewById(R.id.button_start);
+        mButtonCancel = findViewById(R.id.button_cancel);
+
+        mPlain_text_input = findViewById(R.id.plain_text_input);
+        mButtonStart.setOnClickListener((View v) ->  {
+            try {
+                mDtstart = System.currentTimeMillis();
+                mSeries1.resetData(new DataPoint[0]);
+                mMozillaSpeechService.start(getApplicationContext());
+            } catch (Exception e) {
+                Log.d(TAG, e.getLocalizedMessage());
+                e.printStackTrace();
+            }
+        });
+
+        mButtonCancel.setOnClickListener((View v) ->  {
+            try {
+                mMozillaSpeechService.cancel();
+            } catch (Exception e) {
+                Log.d(TAG, e.getLocalizedMessage());
+                e.printStackTrace();
+            }
+        });
+
+        mGraph = findViewById(R.id.graph);
+        mSeries1 = new LineGraphSeries<>(new DataPoint[0]);
+        mGraph.addSeries(mSeries1);
+        mGraph.getViewport().setXAxisBoundsManual(true);
+        mGraph.getViewport().setScalable(true);
+        mGraph.getViewport().setScalableY(true);
+        mGraph.getViewport().setScrollable(true); // enables horizontal scrolling
+        mGraph.getViewport().setScrollableY(true); // enables vertical scrolling
+    }
+
+    public void onRequestPermissionsResult(int requestCode, @NonNull String[] permissions, @NonNull int[] grantResults) {
+        super.onRequestPermissionsResult(requestCode, permissions, grantResults);
+    }
+
+    public void onSpeechStatusChanged(MozillaSpeechService.SpeechState aState, Object aPayload){
+        this.runOnUiThread(() -> {
+            switch (aState) {
+                case DECODING:
+                    mPlain_text_input.append("Decoding... \n");
+                    break;
+                case MIC_ACTIVITY:
+                    long mPointx = System.currentTimeMillis() - mDtstart;
+                    mSeries1.appendData(new DataPoint(Math.round(mPointx) + 1, (double)aPayload * -1), true, 3000);
+                    break;
+                case STT_RESULT:
+                    String message = String.format("Success: %s (%s)", ((STTResult)aPayload).mTranscription, ((STTResult)aPayload).mConfidence);
+                    mPlain_text_input.append(message + "\n");
+                    break;
+                case START_LISTEN:
+                    mPlain_text_input.append("Started to listen\n");
+                    break;
+                case NO_VOICE:
+                    mPlain_text_input.append("No Voice detected\n");
+                    break;
+                case CANCELED:
+                    mPlain_text_input.append("Canceled\n");
+                    break;
+                case ERROR:
+                    mPlain_text_input.append("Error:" + aPayload + " \n");
+                    break;
+                default:
+                    break;
+            }
+        });
+    }
+}
diff --git a/app/src/main/res/drawable-v24/ic_launcher_foreground.xml b/app/src/main/res/drawable-v24/ic_launcher_foreground.xml
new file mode 100644
index 0000000..c7bd21d
--- /dev/null
+++ b/app/src/main/res/drawable-v24/ic_launcher_foreground.xml
@@ -0,0 +1,34 @@
+<vector xmlns:android="http://schemas.android.com/apk/res/android"
+    xmlns:aapt="http://schemas.android.com/aapt"
+    android:width="108dp"
+    android:height="108dp"
+    android:viewportHeight="108"
+    android:viewportWidth="108">
+    <path
+        android:fillType="evenOdd"
+        android:pathData="M32,64C32,64 38.39,52.99 44.13,50.95C51.37,48.37 70.14,49.57 70.14,49.57L108.26,87.69L108,109.01L75.97,107.97L32,64Z"
+        android:strokeColor="#00000000"
+        android:strokeWidth="1">
+        <aapt:attr name="android:fillColor">
+            <gradient
+                android:endX="78.5885"
+                android:endY="90.9159"
+                android:startX="48.7653"
+                android:startY="61.0927"
+                android:type="linear">
+                <item
+                    android:color="#44000000"
+                    android:offset="0.0" />
+                <item
+                    android:color="#00000000"
+                    android:offset="1.0" />
+            </gradient>
+        </aapt:attr>
+    </path>
+    <path
+        android:fillColor="#FFFFFF"
+        android:fillType="nonZero"
+        android:pathData="M66.94,46.02L66.94,46.02C72.44,50.07 76,56.61 76,64L32,64C32,56.61 35.56,50.11 40.98,46.06L36.18,41.19C35.45,40.45 35.45,39.3 36.18,38.56C36.91,37.81 38.05,37.81 38.78,38.56L44.25,44.05C47.18,42.57 50.48,41.71 54,41.71C57.48,41.71 60.78,42.57 63.68,44.05L69.11,38.56C69.84,37.81 70.98,37.81 71.71,38.56C72.44,39.3 72.44,40.45 71.71,41.19L66.94,46.02ZM62.94,56.92C64.08,56.92 65,56.01 65,54.88C65,53.76 64.08,52.85 62.94,52.85C61.8,52.85 60.88,53.76 60.88,54.88C60.88,56.01 61.8,56.92 62.94,56.92ZM45.06,56.92C46.2,56.92 47.13,56.01 47.13,54.88C47.13,53.76 46.2,52.85 45.06,52.85C43.92,52.85 43,53.76 43,54.88C43,56.01 43.92,56.92 45.06,56.92Z"
+        android:strokeColor="#00000000"
+        android:strokeWidth="1" />
+</vector>
diff --git a/app/src/main/res/drawable/ic_launcher_background.xml b/app/src/main/res/drawable/ic_launcher_background.xml
new file mode 100644
index 0000000..d5fccc5
--- /dev/null
+++ b/app/src/main/res/drawable/ic_launcher_background.xml
@@ -0,0 +1,170 @@
+<?xml version="1.0" encoding="utf-8"?>
+<vector xmlns:android="http://schemas.android.com/apk/res/android"
+    android:width="108dp"
+    android:height="108dp"
+    android:viewportHeight="108"
+    android:viewportWidth="108">
+    <path
+        android:fillColor="#26A69A"
+        android:pathData="M0,0h108v108h-108z" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M9,0L9,108"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M19,0L19,108"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M29,0L29,108"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M39,0L39,108"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M49,0L49,108"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M59,0L59,108"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M69,0L69,108"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M79,0L79,108"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M89,0L89,108"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M99,0L99,108"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M0,9L108,9"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M0,19L108,19"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M0,29L108,29"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M0,39L108,39"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M0,49L108,49"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M0,59L108,59"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M0,69L108,69"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M0,79L108,79"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M0,89L108,89"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
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+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
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+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
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+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M19,49L89,49"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
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+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
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+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
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+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
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+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M39,19L39,89"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M49,19L49,89"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M59,19L59,89"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
+        android:pathData="M69,19L69,89"
+        android:strokeColor="#33FFFFFF"
+        android:strokeWidth="0.8" />
+    <path
+        android:fillColor="#00000000"
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diff --git a/app/src/main/res/layout/activity_main.xml b/app/src/main/res/layout/activity_main.xml
new file mode 100644
index 0000000..4c38eef
--- /dev/null
+++ b/app/src/main/res/layout/activity_main.xml
@@ -0,0 +1,52 @@
+<?xml version="1.0" encoding="utf-8"?>
+<android.support.constraint.ConstraintLayout xmlns:android="http://schemas.android.com/apk/res/android"
+    xmlns:app="http://schemas.android.com/apk/res-auto"
+    xmlns:tools="http://schemas.android.com/tools"
+    android:layout_width="match_parent"
+    android:layout_height="match_parent"
+    tools:context="com.mozilla.speechapp.MainActivity">
+    <RelativeLayout
+        android:id="@+id/InnerRelativeLayout"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_alignParentBottom="true" >
+
+    <com.jjoe64.graphview.GraphView
+        android:layout_width="wrap_content"
+        android:layout_height="200dip"
+        android:id="@+id/graph"
+        android:layout_marginStart="15dp"
+        android:layout_marginTop="10dp"
+        android:layout_marginEnd="10dp"
+        />
+
+    <Button
+        android:id="@+id/button_start"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_below="@+id/graph"
+        android:text="Start" />
+
+    <Button
+        android:id="@+id/button_cancel"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_below="@+id/graph"
+        android:layout_toEndOf="@+id/button_start"
+        android:text="Cancel" />
+
+    <EditText
+        android:id="@+id/plain_text_input"
+        android:layout_height="match_parent"
+        android:layout_width="match_parent"
+        android:layout_below="@+id/button_start"
+        android:layout_marginStart="15dp"
+        android:layout_marginTop="10dp"
+        android:layout_marginEnd="10dp"
+        android:layout_marginBottom="10dp"
+        android:singleLine="false"
+        android:gravity="top"
+        android:inputType="textMultiLine"
+        android:enabled="false" />
+    </RelativeLayout>
+</android.support.constraint.ConstraintLayout>
\ No newline at end of file
diff --git a/app/src/main/res/mipmap-anydpi-v26/ic_launcher.xml b/app/src/main/res/mipmap-anydpi-v26/ic_launcher.xml
new file mode 100644
index 0000000..eca70cf
--- /dev/null
+++ b/app/src/main/res/mipmap-anydpi-v26/ic_launcher.xml
@@ -0,0 +1,5 @@
+<?xml version="1.0" encoding="utf-8"?>
+<adaptive-icon xmlns:android="http://schemas.android.com/apk/res/android">
+    <background android:drawable="@drawable/ic_launcher_background" />
+    <foreground android:drawable="@drawable/ic_launcher_foreground" />
+</adaptive-icon>
\ No newline at end of file
diff --git a/app/src/main/res/mipmap-anydpi-v26/ic_launcher_round.xml b/app/src/main/res/mipmap-anydpi-v26/ic_launcher_round.xml
new file mode 100644
index 0000000..eca70cf
--- /dev/null
+++ b/app/src/main/res/mipmap-anydpi-v26/ic_launcher_round.xml
@@ -0,0 +1,5 @@
+<?xml version="1.0" encoding="utf-8"?>
+<adaptive-icon xmlns:android="http://schemas.android.com/apk/res/android">
+    <background android:drawable="@drawable/ic_launcher_background" />
+    <foreground android:drawable="@drawable/ic_launcher_foreground" />
+</adaptive-icon>
\ No newline at end of file
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diff --git a/app/src/main/res/mipmap-hdpi/ic_launcher_round.png b/app/src/main/res/mipmap-hdpi/ic_launcher_round.png
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diff --git a/app/src/main/res/mipmap-xhdpi/ic_launcher_round.png b/app/src/main/res/mipmap-xhdpi/ic_launcher_round.png
new file mode 100644
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diff --git a/app/src/main/res/mipmap-xxhdpi/ic_launcher.png b/app/src/main/res/mipmap-xxhdpi/ic_launcher.png
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diff --git a/app/src/main/res/mipmap-xxhdpi/ic_launcher_round.png b/app/src/main/res/mipmap-xxhdpi/ic_launcher_round.png
new file mode 100644
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diff --git a/app/src/main/res/mipmap-xxxhdpi/ic_launcher.png b/app/src/main/res/mipmap-xxxhdpi/ic_launcher.png
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BIP(Ai

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HcmV?d00001

diff --git a/app/src/main/res/values/colors.xml b/app/src/main/res/values/colors.xml
new file mode 100644
index 0000000..3ab3e9c
--- /dev/null
+++ b/app/src/main/res/values/colors.xml
@@ -0,0 +1,6 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+    <color name="colorPrimary">#3F51B5</color>
+    <color name="colorPrimaryDark">#303F9F</color>
+    <color name="colorAccent">#FF4081</color>
+</resources>
diff --git a/app/src/main/res/values/strings.xml b/app/src/main/res/values/strings.xml
new file mode 100644
index 0000000..193a532
--- /dev/null
+++ b/app/src/main/res/values/strings.xml
@@ -0,0 +1,3 @@
+<resources>
+    <string name="app_name">Speech Module</string>
+</resources>
diff --git a/app/src/main/res/values/styles.xml b/app/src/main/res/values/styles.xml
new file mode 100644
index 0000000..5885930
--- /dev/null
+++ b/app/src/main/res/values/styles.xml
@@ -0,0 +1,11 @@
+<resources>
+
+    <!-- Base application theme. -->
+    <style name="AppTheme" parent="Theme.AppCompat.Light.DarkActionBar">
+        <!-- Customize your theme here. -->
+        <item name="colorPrimary">@color/colorPrimary</item>
+        <item name="colorPrimaryDark">@color/colorPrimaryDark</item>
+        <item name="colorAccent">@color/colorAccent</item>
+    </style>
+
+</resources>
diff --git a/app/src/test/java/com/mozilla/speechmodule/ExampleUnitTest.java b/app/src/test/java/com/mozilla/speechmodule/ExampleUnitTest.java
new file mode 100644
index 0000000..8e233e7
--- /dev/null
+++ b/app/src/test/java/com/mozilla/speechmodule/ExampleUnitTest.java
@@ -0,0 +1,17 @@
+package com.mozilla.speechmodule;
+
+import org.junit.Test;
+
+import static org.junit.Assert.*;
+
+/**
+ * Example local unit test, which will execute on the development machine (host).
+ *
+ * @see <a href="http://d.android.com/tools/testing">Testing documentation</a>
+ */
+public class ExampleUnitTest {
+    @Test
+    public void addition_isCorrect() {
+        assertEquals(4, 2 + 2);
+    }
+}
\ No newline at end of file
diff --git a/build.gradle b/build.gradle
new file mode 100644
index 0000000..263acf5
--- /dev/null
+++ b/build.gradle
@@ -0,0 +1,28 @@
+// Top-level build file where you can add configuration options common to all sub-projects/modules.
+
+buildscript {
+    
+    repositories {
+        google()
+        jcenter()
+    }
+    dependencies {
+        classpath 'com.android.tools.build:gradle:3.1.2'
+        
+
+        // NOTE: Do not place your application dependencies here; they belong
+        // in the individual module build.gradle files
+    }
+}
+
+allprojects {
+    repositories {
+        google()
+        jcenter()
+    }
+}
+
+task clean(type: Delete) {
+    delete rootProject.buildDir
+}
+
diff --git a/gradle.properties b/gradle.properties
new file mode 100644
index 0000000..742196e
--- /dev/null
+++ b/gradle.properties
@@ -0,0 +1,29 @@
+# Project-wide Gradle settings.
+# IDE (e.g. Android Studio) users:
+# Gradle settings configured through the IDE *will override*
+# any settings specified in this file.
+# For more details on how to configure your build environment visit
+# http://www.gradle.org/docs/current/userguide/build_environment.html
+# Specifies the JVM arguments used for the daemon process.
+# The setting is particularly useful for tweaking memory settings.
+org.gradle.jvmargs=-Xmx1536m
+# When configured, Gradle will run in incubating parallel mode.
+# This option should only be used with decoupled projects. More details, visit
+# http://www.gradle.org/docs/current/userguide/multi_project_builds.html#sec:decoupled_projects
+# org.gradle.parallel=true
+
+
+VERSION_NAME=1.0.1
+VERSION_CODE=1
+GROUP=com.github.mozilla
+
+POM_DESCRIPTION=An Android library to Mozilla's speech recognition services
+POM_URL=https://github.com/mozilla/androidspeechlibrary
+POM_SCM_URL=https://github.com/mozilla/androidspeechlibrary
+POM_SCM_CONNECTION=scm:git@github.com:mozilla/androidspeechlibrary.git
+POM_SCM_DEV_CONNECTION=scm:git@github.com:mozilla/androidspeechlibrary.git
+POM_LICENCE_NAME=The Apache Software License, Version 2.0
+POM_LICENCE_URL=http://www.apache.org/licenses/LICENSE-2.0.txt
+POM_LICENCE_DIST=repo
+POM_DEVELOPER_ID=anatal@gmail.com
+POM_DEVELOPER_NAME=Andre Natal
\ No newline at end of file
diff --git a/gradle/wrapper/gradle-wrapper.properties b/gradle/wrapper/gradle-wrapper.properties
new file mode 100644
index 0000000..68db98d
--- /dev/null
+++ b/gradle/wrapper/gradle-wrapper.properties
@@ -0,0 +1,6 @@
+#Thu Aug 23 21:43:36 PDT 2018
+distributionBase=GRADLE_USER_HOME
+distributionPath=wrapper/dists
+zipStoreBase=GRADLE_USER_HOME
+zipStorePath=wrapper/dists
+distributionUrl=https\://services.gradle.org/distributions/gradle-4.4-all.zip
diff --git a/gradlew b/gradlew
new file mode 100755
index 0000000..cccdd3d
--- /dev/null
+++ b/gradlew
@@ -0,0 +1,172 @@
+#!/usr/bin/env sh
+
+##############################################################################
+##
+##  Gradle start up script for UN*X
+##
+##############################################################################
+
+# Attempt to set APP_HOME
+# Resolve links: $0 may be a link
+PRG="$0"
+# Need this for relative symlinks.
+while [ -h "$PRG" ] ; do
+    ls=`ls -ld "$PRG"`
+    link=`expr "$ls" : '.*-> \(.*\)$'`
+    if expr "$link" : '/.*' > /dev/null; then
+        PRG="$link"
+    else
+        PRG=`dirname "$PRG"`"/$link"
+    fi
+done
+SAVED="`pwd`"
+cd "`dirname \"$PRG\"`/" >/dev/null
+APP_HOME="`pwd -P`"
+cd "$SAVED" >/dev/null
+
+APP_NAME="Gradle"
+APP_BASE_NAME=`basename "$0"`
+
+# Add default JVM options here. You can also use JAVA_OPTS and GRADLE_OPTS to pass JVM options to this script.
+DEFAULT_JVM_OPTS=""
+
+# Use the maximum available, or set MAX_FD != -1 to use that value.
+MAX_FD="maximum"
+
+warn () {
+    echo "$*"
+}
+
+die () {
+    echo
+    echo "$*"
+    echo
+    exit 1
+}
+
+# OS specific support (must be 'true' or 'false').
+cygwin=false
+msys=false
+darwin=false
+nonstop=false
+case "`uname`" in
+  CYGWIN* )
+    cygwin=true
+    ;;
+  Darwin* )
+    darwin=true
+    ;;
+  MINGW* )
+    msys=true
+    ;;
+  NONSTOP* )
+    nonstop=true
+    ;;
+esac
+
+CLASSPATH=$APP_HOME/gradle/wrapper/gradle-wrapper.jar
+
+# Determine the Java command to use to start the JVM.
+if [ -n "$JAVA_HOME" ] ; then
+    if [ -x "$JAVA_HOME/jre/sh/java" ] ; then
+        # IBM's JDK on AIX uses strange locations for the executables
+        JAVACMD="$JAVA_HOME/jre/sh/java"
+    else
+        JAVACMD="$JAVA_HOME/bin/java"
+    fi
+    if [ ! -x "$JAVACMD" ] ; then
+        die "ERROR: JAVA_HOME is set to an invalid directory: $JAVA_HOME
+
+Please set the JAVA_HOME variable in your environment to match the
+location of your Java installation."
+    fi
+else
+    JAVACMD="java"
+    which java >/dev/null 2>&1 || die "ERROR: JAVA_HOME is not set and no 'java' command could be found in your PATH.
+
+Please set the JAVA_HOME variable in your environment to match the
+location of your Java installation."
+fi
+
+# Increase the maximum file descriptors if we can.
+if [ "$cygwin" = "false" -a "$darwin" = "false" -a "$nonstop" = "false" ] ; then
+    MAX_FD_LIMIT=`ulimit -H -n`
+    if [ $? -eq 0 ] ; then
+        if [ "$MAX_FD" = "maximum" -o "$MAX_FD" = "max" ] ; then
+            MAX_FD="$MAX_FD_LIMIT"
+        fi
+        ulimit -n $MAX_FD
+        if [ $? -ne 0 ] ; then
+            warn "Could not set maximum file descriptor limit: $MAX_FD"
+        fi
+    else
+        warn "Could not query maximum file descriptor limit: $MAX_FD_LIMIT"
+    fi
+fi
+
+# For Darwin, add options to specify how the application appears in the dock
+if $darwin; then
+    GRADLE_OPTS="$GRADLE_OPTS \"-Xdock:name=$APP_NAME\" \"-Xdock:icon=$APP_HOME/media/gradle.icns\""
+fi
+
+# For Cygwin, switch paths to Windows format before running java
+if $cygwin ; then
+    APP_HOME=`cygpath --path --mixed "$APP_HOME"`
+    CLASSPATH=`cygpath --path --mixed "$CLASSPATH"`
+    JAVACMD=`cygpath --unix "$JAVACMD"`
+
+    # We build the pattern for arguments to be converted via cygpath
+    ROOTDIRSRAW=`find -L / -maxdepth 1 -mindepth 1 -type d 2>/dev/null`
+    SEP=""
+    for dir in $ROOTDIRSRAW ; do
+        ROOTDIRS="$ROOTDIRS$SEP$dir"
+        SEP="|"
+    done
+    OURCYGPATTERN="(^($ROOTDIRS))"
+    # Add a user-defined pattern to the cygpath arguments
+    if [ "$GRADLE_CYGPATTERN" != "" ] ; then
+        OURCYGPATTERN="$OURCYGPATTERN|($GRADLE_CYGPATTERN)"
+    fi
+    # Now convert the arguments - kludge to limit ourselves to /bin/sh
+    i=0
+    for arg in "$@" ; do
+        CHECK=`echo "$arg"|egrep -c "$OURCYGPATTERN" -`
+        CHECK2=`echo "$arg"|egrep -c "^-"`                                 ### Determine if an option
+
+        if [ $CHECK -ne 0 ] && [ $CHECK2 -eq 0 ] ; then                    ### Added a condition
+            eval `echo args$i`=`cygpath --path --ignore --mixed "$arg"`
+        else
+            eval `echo args$i`="\"$arg\""
+        fi
+        i=$((i+1))
+    done
+    case $i in
+        (0) set -- ;;
+        (1) set -- "$args0" ;;
+        (2) set -- "$args0" "$args1" ;;
+        (3) set -- "$args0" "$args1" "$args2" ;;
+        (4) set -- "$args0" "$args1" "$args2" "$args3" ;;
+        (5) set -- "$args0" "$args1" "$args2" "$args3" "$args4" ;;
+        (6) set -- "$args0" "$args1" "$args2" "$args3" "$args4" "$args5" ;;
+        (7) set -- "$args0" "$args1" "$args2" "$args3" "$args4" "$args5" "$args6" ;;
+        (8) set -- "$args0" "$args1" "$args2" "$args3" "$args4" "$args5" "$args6" "$args7" ;;
+        (9) set -- "$args0" "$args1" "$args2" "$args3" "$args4" "$args5" "$args6" "$args7" "$args8" ;;
+    esac
+fi
+
+# Escape application args
+save () {
+    for i do printf %s\\n "$i" | sed "s/'/'\\\\''/g;1s/^/'/;\$s/\$/' \\\\/" ; done
+    echo " "
+}
+APP_ARGS=$(save "$@")
+
+# Collect all arguments for the java command, following the shell quoting and substitution rules
+eval set -- $DEFAULT_JVM_OPTS $JAVA_OPTS $GRADLE_OPTS "\"-Dorg.gradle.appname=$APP_BASE_NAME\"" -classpath "\"$CLASSPATH\"" org.gradle.wrapper.GradleWrapperMain "$APP_ARGS"
+
+# by default we should be in the correct project dir, but when run from Finder on Mac, the cwd is wrong
+if [ "$(uname)" = "Darwin" ] && [ "$HOME" = "$PWD" ]; then
+  cd "$(dirname "$0")"
+fi
+
+exec "$JAVACMD" "$@"
diff --git a/gradlew.bat b/gradlew.bat
new file mode 100644
index 0000000..e95643d
--- /dev/null
+++ b/gradlew.bat
@@ -0,0 +1,84 @@
+@if "%DEBUG%" == "" @echo off
+@rem ##########################################################################
+@rem
+@rem  Gradle startup script for Windows
+@rem
+@rem ##########################################################################
+
+@rem Set local scope for the variables with windows NT shell
+if "%OS%"=="Windows_NT" setlocal
+
+set DIRNAME=%~dp0
+if "%DIRNAME%" == "" set DIRNAME=.
+set APP_BASE_NAME=%~n0
+set APP_HOME=%DIRNAME%
+
+@rem Add default JVM options here. You can also use JAVA_OPTS and GRADLE_OPTS to pass JVM options to this script.
+set DEFAULT_JVM_OPTS=
+
+@rem Find java.exe
+if defined JAVA_HOME goto findJavaFromJavaHome
+
+set JAVA_EXE=java.exe
+%JAVA_EXE% -version >NUL 2>&1
+if "%ERRORLEVEL%" == "0" goto init
+
+echo.
+echo ERROR: JAVA_HOME is not set and no 'java' command could be found in your PATH.
+echo.
+echo Please set the JAVA_HOME variable in your environment to match the
+echo location of your Java installation.
+
+goto fail
+
+:findJavaFromJavaHome
+set JAVA_HOME=%JAVA_HOME:"=%
+set JAVA_EXE=%JAVA_HOME%/bin/java.exe
+
+if exist "%JAVA_EXE%" goto init
+
+echo.
+echo ERROR: JAVA_HOME is set to an invalid directory: %JAVA_HOME%
+echo.
+echo Please set the JAVA_HOME variable in your environment to match the
+echo location of your Java installation.
+
+goto fail
+
+:init
+@rem Get command-line arguments, handling Windows variants
+
+if not "%OS%" == "Windows_NT" goto win9xME_args
+
+:win9xME_args
+@rem Slurp the command line arguments.
+set CMD_LINE_ARGS=
+set _SKIP=2
+
+:win9xME_args_slurp
+if "x%~1" == "x" goto execute
+
+set CMD_LINE_ARGS=%*
+
+:execute
+@rem Setup the command line
+
+set CLASSPATH=%APP_HOME%\gradle\wrapper\gradle-wrapper.jar
+
+@rem Execute Gradle
+"%JAVA_EXE%" %DEFAULT_JVM_OPTS% %JAVA_OPTS% %GRADLE_OPTS% "-Dorg.gradle.appname=%APP_BASE_NAME%" -classpath "%CLASSPATH%" org.gradle.wrapper.GradleWrapperMain %CMD_LINE_ARGS%
+
+:end
+@rem End local scope for the variables with windows NT shell
+if "%ERRORLEVEL%"=="0" goto mainEnd
+
+:fail
+rem Set variable GRADLE_EXIT_CONSOLE if you need the _script_ return code instead of
+rem the _cmd.exe /c_ return code!
+if  not "" == "%GRADLE_EXIT_CONSOLE%" exit 1
+exit /b 1
+
+:mainEnd
+if "%OS%"=="Windows_NT" endlocal
+
+:omega
diff --git a/mozillaspeechlibrary/.gitignore b/mozillaspeechlibrary/.gitignore
new file mode 100644
index 0000000..796b96d
--- /dev/null
+++ b/mozillaspeechlibrary/.gitignore
@@ -0,0 +1 @@
+/build
diff --git a/mozillaspeechlibrary/build.gradle b/mozillaspeechlibrary/build.gradle
new file mode 100644
index 0000000..bfe0d93
--- /dev/null
+++ b/mozillaspeechlibrary/build.gradle
@@ -0,0 +1,49 @@
+apply plugin: 'com.android.library'
+apply from: 'maven-push.gradle'
+
+def versionMajor = 1
+def versionMinor = 0
+def versionPatch = 1
+
+android {
+    compileSdkVersion 25
+    defaultConfig {
+        minSdkVersion 24
+        targetSdkVersion 25
+        versionCode versionMajor * 10000 + versionMinor * 100 + versionPatch
+        versionName "${versionMajor}.${versionMinor}.${versionPatch}"
+
+        testInstrumentationRunner "android.support.test.runner.AndroidJUnitRunner"
+
+    }
+
+    buildTypes {
+        release {
+            minifyEnabled false
+            proguardFiles getDefaultProguardFile('proguard-android.txt'), 'proguard-rules.pro'
+        }
+    }
+
+    externalNativeBuild {
+        ndkBuild {
+            path 'src/main/cpp/Android.mk'
+        }
+    }
+    compileOptions {
+        sourceCompatibility JavaVersion.VERSION_1_8
+        targetCompatibility JavaVersion.VERSION_1_8
+    }
+}
+
+dependencies {
+    implementation fileTree(include: ['*.jar'], dir: 'libs')
+    implementation 'com.loopj.android:android-async-http:1.4.9'
+    implementation 'com.android.support:appcompat-v7:25.4.0'
+    testImplementation 'junit:junit:4.12'
+    androidTestImplementation 'com.android.support.test:runner:1.0.2'
+    androidTestImplementation 'com.android.support.test.espresso:espresso-core:3.0.2'
+    api 'org.gagravarr:vorbis-java-core:0.8'
+    api 'com.github.axet:opus:1.0.2'
+    api 'com.github.axet:jssrc:1.0.2-2'
+    api 'org.apache.commons:commons-math3:3.6.1'
+}
diff --git a/mozillaspeechlibrary/gradle.properties b/mozillaspeechlibrary/gradle.properties
new file mode 100644
index 0000000..9c7a919
--- /dev/null
+++ b/mozillaspeechlibrary/gradle.properties
@@ -0,0 +1,3 @@
+POM_NAME=mozillaspeechlibrary
+POM_ARTIFACT_ID=mozillaspeechlibrary
+POM_PACKAGING=mozillaspeechlibrary
diff --git a/mozillaspeechlibrary/maven-push.gradle b/mozillaspeechlibrary/maven-push.gradle
new file mode 100644
index 0000000..9f186cf
--- /dev/null
+++ b/mozillaspeechlibrary/maven-push.gradle
@@ -0,0 +1,112 @@
+/*
+ * Copyright 2013 Chris Banes
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *     http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+apply plugin: 'maven'
+apply plugin: 'signing'
+
+def isReleaseBuild() {
+    return VERSION_NAME.contains("SNAPSHOT") == false
+}
+
+def getReleaseRepositoryUrl() {
+    return hasProperty('RELEASE_REPOSITORY_URL') ? RELEASE_REPOSITORY_URL
+            : "https://oss.sonatype.org/service/local/staging/deploy/maven2/"
+}
+
+def getSnapshotRepositoryUrl() {
+    return hasProperty('SNAPSHOT_REPOSITORY_URL') ? SNAPSHOT_REPOSITORY_URL
+            : "https://oss.sonatype.org/content/repositories/snapshots/"
+}
+
+def getRepositoryUsername() {
+    return hasProperty('NEXUS_USERNAME') ? NEXUS_USERNAME : ""
+}
+
+def getRepositoryPassword() {
+    return hasProperty('NEXUS_PASSWORD') ? NEXUS_PASSWORD : ""
+}
+
+afterEvaluate { project ->
+    uploadArchives {
+        repositories {
+            mavenDeployer {
+                beforeDeployment { MavenDeployment deployment -> signing.signPom(deployment) }
+
+                pom.groupId = GROUP
+                pom.artifactId = POM_ARTIFACT_ID
+                pom.version = VERSION_NAME
+
+                repository(url: getReleaseRepositoryUrl()) {
+                    authentication(userName: getRepositoryUsername(), password: getRepositoryPassword())
+                }
+                snapshotRepository(url: getSnapshotRepositoryUrl()) {
+                    authentication(userName: getRepositoryUsername(), password: getRepositoryPassword())
+                }
+
+                pom.project {
+                    name POM_NAME
+                    packaging POM_PACKAGING
+                    description POM_DESCRIPTION
+                    url POM_URL
+
+                    scm {
+                        url POM_SCM_URL
+                        connection POM_SCM_CONNECTION
+                        developerConnection POM_SCM_DEV_CONNECTION
+                    }
+
+                    licenses {
+                        license {
+                            name POM_LICENCE_NAME
+                            url POM_LICENCE_URL
+                            distribution POM_LICENCE_DIST
+                        }
+                    }
+
+                    developers {
+                        developer {
+                            id POM_DEVELOPER_ID
+                            name POM_DEVELOPER_NAME
+                        }
+                    }
+                }
+            }
+        }
+    }
+
+    signing {
+        required { isReleaseBuild() && gradle.taskGraph.hasTask("uploadArchives") }
+        sign configurations.archives
+    }
+
+    //task androidJavadocs(type: Javadoc) {
+    //source = android.sourceSets.main.allJava
+    //}
+
+    //task androidJavadocsJar(type: Jar, dependsOn: androidJavadocs) {
+    //classifier = 'javadoc'
+    //from androidJavadocs.destinationDir
+    //}
+
+    task androidSourcesJar(type: Jar) {
+        classifier = 'sources'
+        from android.sourceSets.main.java.sourceFiles
+    }
+
+    artifacts {
+        archives androidSourcesJar
+    }
+}
diff --git a/mozillaspeechlibrary/proguard-rules.pro b/mozillaspeechlibrary/proguard-rules.pro
new file mode 100644
index 0000000..f1b4245
--- /dev/null
+++ b/mozillaspeechlibrary/proguard-rules.pro
@@ -0,0 +1,21 @@
+# Add project specific ProGuard rules here.
+# You can control the set of applied configuration files using the
+# proguardFiles setting in build.gradle.
+#
+# For more details, see
+#   http://developer.android.com/guide/developing/tools/proguard.html
+
+# If your project uses WebView with JS, uncomment the following
+# and specify the fully qualified class name to the JavaScript interface
+# class:
+#-keepclassmembers class fqcn.of.javascript.interface.for.webview {
+#   public *;
+#}
+
+# Uncomment this to preserve the line number information for
+# debugging stack traces.
+#-keepattributes SourceFile,LineNumberTable
+
+# If you keep the line number information, uncomment this to
+# hide the original source file name.
+#-renamesourcefileattribute SourceFile
diff --git a/mozillaspeechlibrary/src/androidTest/java/com/mozilla/speechlibrary/ExampleInstrumentedTest.java b/mozillaspeechlibrary/src/androidTest/java/com/mozilla/speechlibrary/ExampleInstrumentedTest.java
new file mode 100644
index 0000000..b002bb1
--- /dev/null
+++ b/mozillaspeechlibrary/src/androidTest/java/com/mozilla/speechlibrary/ExampleInstrumentedTest.java
@@ -0,0 +1,26 @@
+package com.mozilla.speechlibrary;
+
+import android.content.Context;
+import android.support.test.InstrumentationRegistry;
+import android.support.test.runner.AndroidJUnit4;
+
+import org.junit.Test;
+import org.junit.runner.RunWith;
+
+import static org.junit.Assert.*;
+
+/**
+ * Instrumented test, which will execute on an Android device.
+ *
+ * @see <a href="http://d.android.com/tools/testing">Testing documentation</a>
+ */
+@RunWith(AndroidJUnit4.class)
+public class ExampleInstrumentedTest {
+    @Test
+    public void useAppContext() {
+        // Context of the app under test.
+        Context appContext = InstrumentationRegistry.getTargetContext();
+
+        assertEquals("com.mozilla.speechlibrary.test", appContext.getPackageName());
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/AndroidManifest.xml b/mozillaspeechlibrary/src/main/AndroidManifest.xml
new file mode 100644
index 0000000..a0669de
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/AndroidManifest.xml
@@ -0,0 +1,2 @@
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+    package="com.mozilla.speechlibrary" />
diff --git a/mozillaspeechlibrary/src/main/cpp/Android.mk b/mozillaspeechlibrary/src/main/cpp/Android.mk
new file mode 100644
index 0000000..698b9b0
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/Android.mk
@@ -0,0 +1,37 @@
+
+include $(call all-subdir-makefiles)
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_CFLAGS := -DWEBRTC_ANDROID -D__ARMEL__
+
+LOCAL_MODULE    := webrtc_jni
+LOCAL_SRC_FILES := webrtc_jni.c  \
+  webrtc/common_audio/vad/webrtc_vad.c \
+  webrtc/common_audio/vad/vad_core.c \
+  webrtc/common_audio/vad/vad_filterbank.c \
+  webrtc/common_audio/vad/vad_gmm.c \
+  webrtc/common_audio/vad/vad_sp.c \
+  webrtc/common_audio/signal_processing/spl_init.c \
+  webrtc/common_audio/signal_processing/division_operations.c \
+  webrtc/common_audio/signal_processing/resample_48khz.c \
+  webrtc/common_audio/signal_processing/energy.c \
+  webrtc/common_audio/signal_processing/resample.c \
+  webrtc/common_audio/signal_processing/resample_by_2.c \
+  webrtc/common_audio/signal_processing/resample_fractional.c \
+  webrtc/common_audio/signal_processing/resample_by_2_internal.c \
+  webrtc/common_audio/signal_processing/spl_sqrt.c \
+  webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c \
+  webrtc/common_audio/signal_processing/vector_scaling_operations.c \
+  webrtc/common_audio/signal_processing/real_fft.c \
+  webrtc/common_audio/signal_processing/get_scaling_square.c \
+  webrtc/common_audio/signal_processing/downsample_fast.c \
+  webrtc/common_audio/signal_processing/complex_fft.c \
+  webrtc/common_audio/signal_processing/cross_correlation.c \
+  webrtc/common_audio/signal_processing/complex_bit_reverse.c \
+  webrtc/common_audio/signal_processing/min_max_operations.c
+
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/mozillaspeechlibrary/src/main/cpp/cpu_features_wrapper.h b/mozillaspeechlibrary/src/main/cpp/cpu_features_wrapper.h
new file mode 100644
index 0000000..9f793e3
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/cpu_features_wrapper.h
@@ -0,0 +1,51 @@
+/*
+* 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_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
+#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
+
+#if defined(__cplusplus) || defined(c_plusplus)
+extern "C" {
+#endif
+
+#include <typedefs.h>
+
+// List of features in x86.
+typedef enum {
+  kSSE2,
+  kSSE3
+} CPUFeature;
+
+// List of features in ARM.
+enum {
+  kCPUFeatureARMv7 = (1 << 0),
+  kCPUFeatureVFPv3 = (1 << 1),
+  kCPUFeatureNEON = (1 << 2),
+  kCPUFeatureLDREXSTREX = (1 << 3)
+};
+
+typedef int (*WebRtc_CPUInfo)(CPUFeature feature);
+
+// Returns true if the CPU supports the feature.
+extern WebRtc_CPUInfo WebRtc_GetCPUInfo;
+
+// No CPU feature is available => straight C path.
+extern WebRtc_CPUInfo WebRtc_GetCPUInfoNoASM;
+
+// Return the features in an ARM device.
+// It detects the features in the hardware platform, and returns supported
+// values in the above enum definition as a bitmask.
+extern uint64_t WebRtc_GetCPUFeaturesARM(void);
+
+#if defined(__cplusplus) || defined(c_plusplus)
+} // extern "C"
+#endif
+
+#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/signal_processing_library.h b/mozillaspeechlibrary/src/main/cpp/signal_processing_library.h
new file mode 100644
index 0000000..54a8623
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/signal_processing_library.h
@@ -0,0 +1,1686 @@
+/*
+ *  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 all of the fix point signal processing library (SPL) function
+ * descriptions and declarations.
+ * For specific function calls, see bottom of file.
+ */
+
+#ifndef WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+#define WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+
+#include <string.h>
+#include "typedefs.h"
+
+#ifdef ARM_WINM
+#include <Armintr.h> // intrinsic file for windows mobile
+#endif
+
+// Macros specific for the fixed point implementation
+#define WEBRTC_SPL_WORD16_MAX       32767
+#define WEBRTC_SPL_WORD16_MIN       -32768
+#define WEBRTC_SPL_WORD32_MAX       (WebRtc_Word32)0x7fffffff
+#define WEBRTC_SPL_WORD32_MIN       (WebRtc_Word32)0x80000000
+#define WEBRTC_SPL_MAX_LPC_ORDER    14
+#define WEBRTC_SPL_MAX_SEED_USED    0x80000000L
+#define WEBRTC_SPL_MIN(A, B)        (A < B ? A : B) // Get min value
+#define WEBRTC_SPL_MAX(A, B)        (A > B ? A : B) // Get max value
+#define WEBRTC_SPL_ABS_W16(a) \
+    (((WebRtc_Word16)a >= 0) ? ((WebRtc_Word16)a) : -((WebRtc_Word16)a))
+#define WEBRTC_SPL_ABS_W32(a) \
+    (((WebRtc_Word32)a >= 0) ? ((WebRtc_Word32)a) : -((WebRtc_Word32)a))
+
+#if (defined WEBRTC_TARGET_PC)||(defined __TARGET_XSCALE)
+#define WEBRTC_SPL_GET_BYTE(a, nr)  (((WebRtc_Word8 *)a)[nr])
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+    (((WebRtc_Word8 *)d_ptr)[index] = (val))
+#elif defined WEBRTC_BIG_ENDIAN
+#define WEBRTC_SPL_GET_BYTE(a, nr) \
+    ((((WebRtc_Word16 *)a)[nr >> 1]) >> (((nr + 1) & 0x1) * 8) & 0x00ff)
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+    ((WebRtc_Word16 *)d_ptr)[index >> 1] = \
+    ((((WebRtc_Word16 *)d_ptr)[index >> 1]) \
+    & (0x00ff << (8 * ((index) & 0x1)))) | (val << (8 * ((index + 1) & 0x1)))
+#else
+#define WEBRTC_SPL_GET_BYTE(a,nr) \
+    ((((WebRtc_Word16 *)(a))[(nr) >> 1]) >> (((nr) & 0x1) * 8) & 0x00ff)
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+    ((WebRtc_Word16 *)(d_ptr))[(index) >> 1] = \
+    ((((WebRtc_Word16 *)(d_ptr))[(index) >> 1]) \
+    & (0x00ff << (8 * (((index) + 1) & 0x1)))) | \
+    ((val) << (8 * ((index) & 0x1)))
+#endif
+
+#define WEBRTC_SPL_MUL(a, b) \
+    ((WebRtc_Word32) ((WebRtc_Word32)(a) * (WebRtc_Word32)(b)))
+#define WEBRTC_SPL_UMUL(a, b) \
+    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)))
+#define WEBRTC_SPL_UMUL_RSFT16(a, b) \
+    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)) >> 16)
+#define WEBRTC_SPL_UMUL_16_16(a, b) \
+    ((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b))
+#define WEBRTC_SPL_UMUL_16_16_RSFT16(a, b) \
+    (((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b)) >> 16)
+#define WEBRTC_SPL_UMUL_32_16(a, b) \
+    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)))
+#define WEBRTC_SPL_UMUL_32_16_RSFT16(a, b) \
+    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)) >> 16)
+#define WEBRTC_SPL_MUL_16_U16(a, b) \
+    ((WebRtc_Word32)(WebRtc_Word16)(a) * (WebRtc_UWord16)(b))
+#define WEBRTC_SPL_DIV(a, b) \
+    ((WebRtc_Word32) ((WebRtc_Word32)(a) / (WebRtc_Word32)(b)))
+#define WEBRTC_SPL_UDIV(a, b) \
+    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) / (WebRtc_UWord32)(b)))
+
+#ifndef WEBRTC_ARCH_ARM_V7A
+// For ARMv7 platforms, these are inline functions in spl_inl_armv7.h
+#define WEBRTC_SPL_MUL_16_16(a, b) \
+    ((WebRtc_Word32) (((WebRtc_Word16)(a)) * ((WebRtc_Word16)(b))))
+#define WEBRTC_SPL_MUL_16_32_RSFT16(a, b) \
+    (WEBRTC_SPL_MUL_16_16(a, b >> 16) \
+     + ((WEBRTC_SPL_MUL_16_16(a, (b & 0xffff) >> 1) + 0x4000) >> 15))
+#define WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, b32) \
+    ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(a32a, b32) \
+    + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32) >> 16)))
+#define WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, b32) \
+    ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(( \
+    (WebRtc_Word16)(a32 >> 16)), b32) + \
+    (WEBRTC_SPL_MUL_16_32_RSFT16(( \
+    (WebRtc_Word16)((a32 & 0x0000FFFF) >> 1)), b32) >> 15)))
+#endif
+
+#define WEBRTC_SPL_MUL_16_32_RSFT11(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 5) \
+    + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x0200) >> 10))
+#define WEBRTC_SPL_MUL_16_32_RSFT14(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 2) \
+    + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x1000) >> 13))
+#define WEBRTC_SPL_MUL_16_32_RSFT15(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 1) \
+    + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x2000) >> 14))
+
+#ifdef ARM_WINM
+#define WEBRTC_SPL_MUL_16_16(a, b) \
+    _SmulLo_SW_SL((WebRtc_Word16)(a), (WebRtc_Word16)(b))
+#endif
+
+#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
+    (WEBRTC_SPL_MUL_16_16(a, b) >> (c))
+
+#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c) \
+    ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) \
+                                  (((WebRtc_Word32)1) << ((c) - 1)))) >> (c))
+#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) (1 << 14))) >> 15)
+
+// C + the 32 most significant bits of A * B
+#define WEBRTC_SPL_SCALEDIFF32(A, B, C) \
+    (C + (B >> 16) * A + (((WebRtc_UWord32)(0x0000FFFF & B) * A) >> 16))
+
+#define WEBRTC_SPL_ADD_SAT_W32(a, b)    WebRtcSpl_AddSatW32(a, b)
+#define WEBRTC_SPL_SAT(a, b, c)         (b > a ? a : b < c ? c : b)
+#define WEBRTC_SPL_MUL_32_16(a, b)      ((a) * (b))
+
+#define WEBRTC_SPL_SUB_SAT_W32(a, b)    WebRtcSpl_SubSatW32(a, b)
+#define WEBRTC_SPL_ADD_SAT_W16(a, b)    WebRtcSpl_AddSatW16(a, b)
+#define WEBRTC_SPL_SUB_SAT_W16(a, b)    WebRtcSpl_SubSatW16(a, b)
+
+// We cannot do casting here due to signed/unsigned problem
+#define WEBRTC_SPL_IS_NEG(a)            ((a) & 0x80000000)
+// Shifting with negative numbers allowed
+// Positive means left shift
+#define WEBRTC_SPL_SHIFT_W16(x, c) \
+    (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
+#define WEBRTC_SPL_SHIFT_W32(x, c) \
+    (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
+
+// Shifting with negative numbers not allowed
+// We cannot do casting here due to signed/unsigned problem
+#define WEBRTC_SPL_RSHIFT_W16(x, c)     ((x) >> (c))
+#define WEBRTC_SPL_LSHIFT_W16(x, c)     ((x) << (c))
+#define WEBRTC_SPL_RSHIFT_W32(x, c)     ((x) >> (c))
+#define WEBRTC_SPL_LSHIFT_W32(x, c)     ((x) << (c))
+
+#define WEBRTC_SPL_RSHIFT_U16(x, c)     ((WebRtc_UWord16)(x) >> (c))
+#define WEBRTC_SPL_LSHIFT_U16(x, c)     ((WebRtc_UWord16)(x) << (c))
+#define WEBRTC_SPL_RSHIFT_U32(x, c)     ((WebRtc_UWord32)(x) >> (c))
+#define WEBRTC_SPL_LSHIFT_U32(x, c)     ((WebRtc_UWord32)(x) << (c))
+
+#define WEBRTC_SPL_VNEW(t, n)           (t *) malloc (sizeof (t) * (n))
+#define WEBRTC_SPL_FREE                 free
+
+#define WEBRTC_SPL_RAND(a) \
+    ((WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT((a), 18816, 7) & 0x00007fff))
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+#define WEBRTC_SPL_MEMCPY_W8(v1, v2, length) \
+   memcpy(v1, v2, (length) * sizeof(char))
+#define WEBRTC_SPL_MEMCPY_W16(v1, v2, length) \
+   memcpy(v1, v2, (length) * sizeof(WebRtc_Word16))
+
+#define WEBRTC_SPL_MEMMOVE_W16(v1, v2, length) \
+   memmove(v1, v2, (length) * sizeof(WebRtc_Word16))
+
+// inline functions:
+#include "spl_inl.h"
+
+// Get SPL Version
+WebRtc_Word16 WebRtcSpl_get_version(char* version,
+                                    WebRtc_Word16 length_in_bytes);
+
+int WebRtcSpl_GetScalingSquare(WebRtc_Word16* in_vector,
+                               int in_vector_length,
+                               int times);
+
+// Copy and set operations. Implementation in copy_set_operations.c.
+// Descriptions at bottom of file.
+void WebRtcSpl_MemSetW16(WebRtc_Word16* vector,
+                         WebRtc_Word16 set_value,
+                         int vector_length);
+void WebRtcSpl_MemSetW32(WebRtc_Word32* vector,
+                         WebRtc_Word32 set_value,
+                         int vector_length);
+void WebRtcSpl_MemCpyReversedOrder(WebRtc_Word16* out_vector,
+                                   WebRtc_Word16* in_vector,
+                                   int vector_length);
+WebRtc_Word16 WebRtcSpl_CopyFromEndW16(G_CONST WebRtc_Word16* in_vector,
+                                       WebRtc_Word16 in_vector_length,
+                                       WebRtc_Word16 samples,
+                                       WebRtc_Word16* out_vector);
+WebRtc_Word16 WebRtcSpl_ZerosArrayW16(WebRtc_Word16* vector,
+                                      WebRtc_Word16 vector_length);
+WebRtc_Word16 WebRtcSpl_ZerosArrayW32(WebRtc_Word32* vector,
+                                      WebRtc_Word16 vector_length);
+WebRtc_Word16 WebRtcSpl_OnesArrayW16(WebRtc_Word16* vector,
+                                     WebRtc_Word16 vector_length);
+WebRtc_Word16 WebRtcSpl_OnesArrayW32(WebRtc_Word32* vector,
+                                     WebRtc_Word16 vector_length);
+// End: Copy and set operations.
+
+// Minimum and maximum operations. Implementation in min_max_operations.c.
+// Descriptions at bottom of file.
+WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16* vector,
+                                       WebRtc_Word16 length);
+WebRtc_Word32 WebRtcSpl_MaxAbsValueW32(G_CONST WebRtc_Word32* vector,
+                                       WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MinValueW16(G_CONST WebRtc_Word16* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word32 WebRtcSpl_MinValueW32(G_CONST WebRtc_Word32* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MaxValueW16(G_CONST WebRtc_Word16* vector,
+                                    WebRtc_Word16 length);
+
+WebRtc_Word16 WebRtcSpl_MaxAbsIndexW16(G_CONST WebRtc_Word16* vector,
+                                       WebRtc_Word16 length);
+WebRtc_Word32 WebRtcSpl_MaxValueW32(G_CONST WebRtc_Word32* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MinIndexW16(G_CONST WebRtc_Word16* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MinIndexW32(G_CONST WebRtc_Word32* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MaxIndexW16(G_CONST WebRtc_Word16* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MaxIndexW32(G_CONST WebRtc_Word32* vector,
+                                    WebRtc_Word16 length);
+// End: Minimum and maximum operations.
+
+// Vector scaling operations. Implementation in vector_scaling_operations.c.
+// Description at bottom of file.
+void WebRtcSpl_VectorBitShiftW16(WebRtc_Word16* out_vector,
+                                 WebRtc_Word16 vector_length,
+                                 G_CONST WebRtc_Word16* in_vector,
+                                 WebRtc_Word16 right_shifts);
+void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32* out_vector,
+                                 WebRtc_Word16 vector_length,
+                                 G_CONST WebRtc_Word32* in_vector,
+                                 WebRtc_Word16 right_shifts);
+void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16* out_vector,
+                                      WebRtc_Word16 vector_length,
+                                      G_CONST WebRtc_Word32* in_vector,
+                                      WebRtc_Word16 right_shifts);
+
+void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16* in_vector,
+                           WebRtc_Word16* out_vector,
+                           WebRtc_Word16 gain,
+                           WebRtc_Word16 vector_length,
+                           WebRtc_Word16 right_shifts);
+void WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16* in_vector,
+                                  WebRtc_Word16* out_vector,
+                                  WebRtc_Word16 gain,
+                                  WebRtc_Word16 vector_length,
+                                  WebRtc_Word16 right_shifts);
+void WebRtcSpl_ScaleAndAddVectors(G_CONST WebRtc_Word16* in_vector1,
+                                  WebRtc_Word16 gain1, int right_shifts1,
+                                  G_CONST WebRtc_Word16* in_vector2,
+                                  WebRtc_Word16 gain2, int right_shifts2,
+                                  WebRtc_Word16* out_vector,
+                                  int vector_length);
+// End: Vector scaling operations.
+
+// iLBC specific functions. Implementations in ilbc_specific_functions.c.
+// Description at bottom of file.
+void WebRtcSpl_ScaleAndAddVectorsWithRound(WebRtc_Word16* in_vector1,
+                                           WebRtc_Word16 scale1,
+                                           WebRtc_Word16* in_vector2,
+                                           WebRtc_Word16 scale2,
+                                           WebRtc_Word16 right_shifts,
+                                           WebRtc_Word16* out_vector,
+                                           WebRtc_Word16 vector_length);
+void WebRtcSpl_ReverseOrderMultArrayElements(WebRtc_Word16* out_vector,
+                                             G_CONST WebRtc_Word16* in_vector,
+                                             G_CONST WebRtc_Word16* window,
+                                             WebRtc_Word16 vector_length,
+                                             WebRtc_Word16 right_shifts);
+void WebRtcSpl_ElementwiseVectorMult(WebRtc_Word16* out_vector,
+                                     G_CONST WebRtc_Word16* in_vector,
+                                     G_CONST WebRtc_Word16* window,
+                                     WebRtc_Word16 vector_length,
+                                     WebRtc_Word16 right_shifts);
+void WebRtcSpl_AddVectorsAndShift(WebRtc_Word16* out_vector,
+                                  G_CONST WebRtc_Word16* in_vector1,
+                                  G_CONST WebRtc_Word16* in_vector2,
+                                  WebRtc_Word16 vector_length,
+                                  WebRtc_Word16 right_shifts);
+void WebRtcSpl_AddAffineVectorToVector(WebRtc_Word16* out_vector,
+                                       WebRtc_Word16* in_vector,
+                                       WebRtc_Word16 gain,
+                                       WebRtc_Word32 add_constant,
+                                       WebRtc_Word16 right_shifts,
+                                       int vector_length);
+void WebRtcSpl_AffineTransformVector(WebRtc_Word16* out_vector,
+                                     WebRtc_Word16* in_vector,
+                                     WebRtc_Word16 gain,
+                                     WebRtc_Word32 add_constant,
+                                     WebRtc_Word16 right_shifts,
+                                     int vector_length);
+// End: iLBC specific functions.
+
+// Signal processing operations. Descriptions at bottom of this file.
+int WebRtcSpl_AutoCorrelation(G_CONST WebRtc_Word16* vector,
+                              int vector_length, int order,
+                              WebRtc_Word32* result_vector,
+                              int* scale);
+WebRtc_Word16 WebRtcSpl_LevinsonDurbin(WebRtc_Word32* auto_corr,
+                                       WebRtc_Word16* lpc_coef,
+                                       WebRtc_Word16* refl_coef,
+                                       WebRtc_Word16 order);
+void WebRtcSpl_ReflCoefToLpc(G_CONST WebRtc_Word16* refl_coef,
+                             int use_order,
+                             WebRtc_Word16* lpc_coef);
+void WebRtcSpl_LpcToReflCoef(WebRtc_Word16* lpc_coef,
+                             int use_order,
+                             WebRtc_Word16* refl_coef);
+void WebRtcSpl_AutoCorrToReflCoef(G_CONST WebRtc_Word32* auto_corr,
+                                  int use_order,
+                                  WebRtc_Word16* refl_coef);
+void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_corr,
+                                WebRtc_Word16* vector1,
+                                WebRtc_Word16* vector2,
+                                WebRtc_Word16 dim_vector,
+                                WebRtc_Word16 dim_cross_corr,
+                                WebRtc_Word16 right_shifts,
+                                WebRtc_Word16 step_vector2);
+void WebRtcSpl_GetHanningWindow(WebRtc_Word16* window, WebRtc_Word16 size);
+void WebRtcSpl_SqrtOfOneMinusXSquared(WebRtc_Word16* in_vector,
+                                      int vector_length,
+                                      WebRtc_Word16* out_vector);
+// End: Signal processing operations.
+
+// Randomization functions. Implementations collected in randomization_functions.c and
+// descriptions at bottom of this file.
+WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32* seed);
+WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32* seed);
+WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32* seed);
+WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
+                                   WebRtc_Word16 vector_length,
+                                   WebRtc_UWord32* seed);
+// End: Randomization functions.
+
+// Math functions
+WebRtc_Word32 WebRtcSpl_Sqrt(WebRtc_Word32 value);
+WebRtc_Word32 WebRtcSpl_SqrtFloor(WebRtc_Word32 value);
+
+// Divisions. Implementations collected in division_operations.c and
+// descriptions at bottom of this file.
+WebRtc_UWord32 WebRtcSpl_DivU32U16(WebRtc_UWord32 num, WebRtc_UWord16 den);
+WebRtc_Word32 WebRtcSpl_DivW32W16(WebRtc_Word32 num, WebRtc_Word16 den);
+WebRtc_Word16 WebRtcSpl_DivW32W16ResW16(WebRtc_Word32 num, WebRtc_Word16 den);
+WebRtc_Word32 WebRtcSpl_DivResultInQ31(WebRtc_Word32 num, WebRtc_Word32 den);
+WebRtc_Word32 WebRtcSpl_DivW32HiLow(WebRtc_Word32 num, WebRtc_Word16 den_hi,
+                                    WebRtc_Word16 den_low);
+// End: Divisions.
+
+WebRtc_Word32 WebRtcSpl_Energy(WebRtc_Word16* vector,
+                               int vector_length,
+                               int* scale_factor);
+
+WebRtc_Word32 WebRtcSpl_DotProductWithScale(WebRtc_Word16* vector1,
+                                            WebRtc_Word16* vector2,
+                                            int vector_length,
+                                            int scaling);
+
+// Filter operations.
+int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* ar_coef, int ar_coef_length,
+                       G_CONST WebRtc_Word16* in_vector, int in_vector_length,
+                       WebRtc_Word16* filter_state, int filter_state_length,
+                       WebRtc_Word16* filter_state_low,
+                       int filter_state_low_length, WebRtc_Word16* out_vector,
+                       WebRtc_Word16* out_vector_low, int out_vector_low_length);
+
+void WebRtcSpl_FilterMAFastQ12(WebRtc_Word16* in_vector,
+                               WebRtc_Word16* out_vector,
+                               WebRtc_Word16* ma_coef,
+                               WebRtc_Word16 ma_coef_length,
+                               WebRtc_Word16 vector_length);
+void WebRtcSpl_FilterARFastQ12(WebRtc_Word16* in_vector,
+                               WebRtc_Word16* out_vector,
+                               WebRtc_Word16* ar_coef,
+                               WebRtc_Word16 ar_coef_length,
+                               WebRtc_Word16 vector_length);
+int WebRtcSpl_DownsampleFast(WebRtc_Word16* in_vector,
+                             WebRtc_Word16 in_vector_length,
+                             WebRtc_Word16* out_vector,
+                             WebRtc_Word16 out_vector_length,
+                             WebRtc_Word16* ma_coef,
+                             WebRtc_Word16 ma_coef_length,
+                             WebRtc_Word16 factor,
+                             WebRtc_Word16 delay);
+// End: Filter operations.
+
+// FFT operations
+int WebRtcSpl_ComplexFFT(WebRtc_Word16 vector[], int stages, int mode);
+int WebRtcSpl_ComplexIFFT(WebRtc_Word16 vector[], int stages, int mode);
+void WebRtcSpl_ComplexBitReverse(WebRtc_Word16 vector[], int stages);
+// End: FFT operations
+
+/************************************************************
+ *
+ * RESAMPLING FUNCTIONS AND THEIR STRUCTS ARE DEFINED BELOW
+ *
+ ************************************************************/
+
+/*******************************************************************
+ * resample.c
+ *
+ * Includes the following resampling combinations
+ * 22 kHz -> 16 kHz
+ * 16 kHz -> 22 kHz
+ * 22 kHz ->  8 kHz
+ *  8 kHz -> 22 kHz
+ *
+ ******************************************************************/
+
+// state structure for 22 -> 16 resampler
+typedef struct
+{
+    WebRtc_Word32 S_22_44[8];
+    WebRtc_Word32 S_44_32[8];
+    WebRtc_Word32 S_32_16[8];
+} WebRtcSpl_State22khzTo16khz;
+
+void WebRtcSpl_Resample22khzTo16khz(const WebRtc_Word16* in,
+                                    WebRtc_Word16* out,
+                                    WebRtcSpl_State22khzTo16khz* state,
+                                    WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state);
+
+// state structure for 16 -> 22 resampler
+typedef struct
+{
+    WebRtc_Word32 S_16_32[8];
+    WebRtc_Word32 S_32_22[8];
+} WebRtcSpl_State16khzTo22khz;
+
+void WebRtcSpl_Resample16khzTo22khz(const WebRtc_Word16* in,
+                                    WebRtc_Word16* out,
+                                    WebRtcSpl_State16khzTo22khz* state,
+                                    WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state);
+
+// state structure for 22 -> 8 resampler
+typedef struct
+{
+    WebRtc_Word32 S_22_22[16];
+    WebRtc_Word32 S_22_16[8];
+    WebRtc_Word32 S_16_8[8];
+} WebRtcSpl_State22khzTo8khz;
+
+void WebRtcSpl_Resample22khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State22khzTo8khz* state,
+                                   WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state);
+
+// state structure for 8 -> 22 resampler
+typedef struct
+{
+    WebRtc_Word32 S_8_16[8];
+    WebRtc_Word32 S_16_11[8];
+    WebRtc_Word32 S_11_22[8];
+} WebRtcSpl_State8khzTo22khz;
+
+void WebRtcSpl_Resample8khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State8khzTo22khz* state,
+                                   WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state);
+
+/*******************************************************************
+ * resample_fractional.c
+ * Functions for internal use in the other resample functions
+ *
+ * Includes the following resampling combinations
+ * 48 kHz -> 32 kHz
+ * 32 kHz -> 24 kHz
+ * 44 kHz -> 32 kHz
+ *
+ ******************************************************************/
+
+void WebRtcSpl_Resample48khzTo32khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
+                                    const WebRtc_Word32 K);
+
+void WebRtcSpl_Resample32khzTo24khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
+                                    const WebRtc_Word32 K);
+
+void WebRtcSpl_Resample44khzTo32khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
+                                    const WebRtc_Word32 K);
+
+/*******************************************************************
+ * resample_48khz.c
+ *
+ * Includes the following resampling combinations
+ * 48 kHz -> 16 kHz
+ * 16 kHz -> 48 kHz
+ * 48 kHz ->  8 kHz
+ *  8 kHz -> 48 kHz
+ *
+ ******************************************************************/
+
+typedef struct
+{
+    WebRtc_Word32 S_48_48[16];
+    WebRtc_Word32 S_48_32[8];
+    WebRtc_Word32 S_32_16[8];
+} WebRtcSpl_State48khzTo16khz;
+
+void WebRtcSpl_Resample48khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                    WebRtcSpl_State48khzTo16khz* state,
+                                    WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state);
+
+typedef struct
+{
+    WebRtc_Word32 S_16_32[8];
+    WebRtc_Word32 S_32_24[8];
+    WebRtc_Word32 S_24_48[8];
+} WebRtcSpl_State16khzTo48khz;
+
+void WebRtcSpl_Resample16khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                    WebRtcSpl_State16khzTo48khz* state,
+                                    WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state);
+
+typedef struct
+{
+    WebRtc_Word32 S_48_24[8];
+    WebRtc_Word32 S_24_24[16];
+    WebRtc_Word32 S_24_16[8];
+    WebRtc_Word32 S_16_8[8];
+} WebRtcSpl_State48khzTo8khz;
+
+void WebRtcSpl_Resample48khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State48khzTo8khz* state,
+                                   WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state);
+
+typedef struct
+{
+    WebRtc_Word32 S_8_16[8];
+    WebRtc_Word32 S_16_12[8];
+    WebRtc_Word32 S_12_24[8];
+    WebRtc_Word32 S_24_48[8];
+} WebRtcSpl_State8khzTo48khz;
+
+void WebRtcSpl_Resample8khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State8khzTo48khz* state,
+                                   WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state);
+
+/*******************************************************************
+ * resample_by_2.c
+ *
+ * Includes down and up sampling by a factor of two.
+ *
+ ******************************************************************/
+
+void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
+                             WebRtc_Word16* out, WebRtc_Word32* filtState);
+
+void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len, WebRtc_Word16* out,
+                           WebRtc_Word32* filtState);
+
+/************************************************************
+ * END OF RESAMPLING FUNCTIONS
+ ************************************************************/
+void WebRtcSpl_AnalysisQMF(const WebRtc_Word16* in_data,
+                           WebRtc_Word16* low_band,
+                           WebRtc_Word16* high_band,
+                           WebRtc_Word32* filter_state1,
+                           WebRtc_Word32* filter_state2);
+void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band,
+                            const WebRtc_Word16* high_band,
+                            WebRtc_Word16* out_data,
+                            WebRtc_Word32* filter_state1,
+                            WebRtc_Word32* filter_state2);
+
+#ifdef __cplusplus
+}
+#endif // __cplusplus
+#endif // WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+
+//
+// WebRtcSpl_AddSatW16(...)
+// WebRtcSpl_AddSatW32(...)
+//
+// Returns the result of a saturated 16-bit, respectively 32-bit, addition of
+// the numbers specified by the |var1| and |var2| parameters.
+//
+// Input:
+//      - var1      : Input variable 1
+//      - var2      : Input variable 2
+//
+// Return value     : Added and saturated value
+//
+
+//
+// WebRtcSpl_SubSatW16(...)
+// WebRtcSpl_SubSatW32(...)
+//
+// Returns the result of a saturated 16-bit, respectively 32-bit, subtraction
+// of the numbers specified by the |var1| and |var2| parameters.
+//
+// Input:
+//      - var1      : Input variable 1
+//      - var2      : Input variable 2
+//
+// Returned value   : Subtracted and saturated value
+//
+
+//
+// WebRtcSpl_GetSizeInBits(...)
+//
+// Returns the # of bits that are needed at the most to represent the number
+// specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bits needed to represent |value|
+//
+
+//
+// WebRtcSpl_NormW32(...)
+//
+// Norm returns the # of left shifts required to 32-bit normalize the 32-bit
+// signed number specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_NormW16(...)
+//
+// Norm returns the # of left shifts required to 16-bit normalize the 16-bit
+// signed number specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_NormU32(...)
+//
+// Norm returns the # of left shifts required to 32-bit normalize the unsigned
+// 32-bit number specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_GetScalingSquare(...)
+//
+// Returns the # of bits required to scale the samples specified in the
+// |in_vector| parameter so that, if the squares of the samples are added the
+// # of times specified by the |times| parameter, the 32-bit addition will not
+// overflow (result in WebRtc_Word32).
+//
+// Input:
+//      - in_vector         : Input vector to check scaling on
+//      - in_vector_length  : Samples in |in_vector|
+//      - times             : Number of additions to be performed
+//
+// Return value             : Number of right bit shifts needed to avoid
+//                            overflow in the addition calculation
+//
+
+//
+// WebRtcSpl_MemSetW16(...)
+//
+// Sets all the values in the WebRtc_Word16 vector |vector| of length
+// |vector_length| to the specified value |set_value|
+//
+// Input:
+//      - vector        : Pointer to the WebRtc_Word16 vector
+//      - set_value     : Value specified
+//      - vector_length : Length of vector
+//
+
+//
+// WebRtcSpl_MemSetW32(...)
+//
+// Sets all the values in the WebRtc_Word32 vector |vector| of length
+// |vector_length| to the specified value |set_value|
+//
+// Input:
+//      - vector        : Pointer to the WebRtc_Word16 vector
+//      - set_value     : Value specified
+//      - vector_length : Length of vector
+//
+
+//
+// WebRtcSpl_MemCpyReversedOrder(...)
+//
+// Copies all the values from the source WebRtc_Word16 vector |in_vector| to a
+// destination WebRtc_Word16 vector |out_vector|. It is done in reversed order,
+// meaning that the first sample of |in_vector| is copied to the last sample of
+// the |out_vector|. The procedure continues until the last sample of
+// |in_vector| has been copied to the first sample of |out_vector|. This
+// creates a reversed vector. Used in e.g. prediction in iLBC.
+//
+// Input:
+//      - in_vector     : Pointer to the first sample in a WebRtc_Word16 vector
+//                        of length |length|
+//      - vector_length : Number of elements to copy
+//
+// Output:
+//      - out_vector    : Pointer to the last sample in a WebRtc_Word16 vector
+//                        of length |length|
+//
+
+//
+// WebRtcSpl_CopyFromEndW16(...)
+//
+// Copies the rightmost |samples| of |in_vector| (of length |in_vector_length|)
+// to the vector |out_vector|.
+//
+// Input:
+//      - in_vector         : Input vector
+//      - in_vector_length  : Number of samples in |in_vector|
+//      - samples           : Number of samples to extract (from right side)
+//                            from |in_vector|
+//
+// Output:
+//      - out_vector        : Vector with the requested samples
+//
+// Return value             : Number of copied samples in |out_vector|
+//
+
+//
+// WebRtcSpl_ZerosArrayW16(...)
+// WebRtcSpl_ZerosArrayW32(...)
+//
+// Inserts the value "zero" in all positions of a w16 and a w32 vector
+// respectively.
+//
+// Input:
+//      - vector_length : Number of samples in vector
+//
+// Output:
+//      - vector        : Vector containing all zeros
+//
+// Return value         : Number of samples in vector
+//
+
+//
+// WebRtcSpl_OnesArrayW16(...)
+// WebRtcSpl_OnesArrayW32(...)
+//
+// Inserts the value "one" in all positions of a w16 and a w32 vector
+// respectively.
+//
+// Input:
+//      - vector_length : Number of samples in vector
+//
+// Output:
+//      - vector        : Vector containing all ones
+//
+// Return value         : Number of samples in vector
+//
+
+//
+// WebRtcSpl_MinValueW16(...)
+// WebRtcSpl_MinValueW32(...)
+//
+// Returns the minimum value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Minimum sample value in vector
+//
+
+//
+// WebRtcSpl_MaxValueW16(...)
+// WebRtcSpl_MaxValueW32(...)
+//
+// Returns the maximum value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Maximum sample value in vector
+//
+
+//
+// WebRtcSpl_MaxAbsValueW16(...)
+// WebRtcSpl_MaxAbsValueW32(...)
+//
+// Returns the largest absolute value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Maximum absolute value in vector
+//
+
+//
+// WebRtcSpl_MaxAbsIndexW16(...)
+//
+// Returns the vector index to the largest absolute value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Index to maximum absolute value in vector
+//
+
+//
+// WebRtcSpl_MinIndexW16(...)
+// WebRtcSpl_MinIndexW32(...)
+//
+// Returns the vector index to the minimum sample value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Index to minimum sample value in vector
+//
+
+//
+// WebRtcSpl_MaxIndexW16(...)
+// WebRtcSpl_MaxIndexW32(...)
+//
+// Returns the vector index to the maximum sample value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Index to maximum sample value in vector
+//
+
+//
+// WebRtcSpl_VectorBitShiftW16(...)
+// WebRtcSpl_VectorBitShiftW32(...)
+//
+// Bit shifts all the values in a vector up or downwards. Different calls for
+// WebRtc_Word16 and WebRtc_Word32 vectors respectively.
+//
+// Input:
+//      - vector_length : Length of vector
+//      - in_vector     : Pointer to the vector that should be bit shifted
+//      - right_shifts  : Number of right bit shifts (negative value gives left
+//                        shifts)
+//
+// Output:
+//      - out_vector    : Pointer to the result vector (can be the same as
+//                        |in_vector|)
+//
+
+//
+// WebRtcSpl_VectorBitShiftW32ToW16(...)
+//
+// Bit shifts all the values in a WebRtc_Word32 vector up or downwards and
+// stores the result as a WebRtc_Word16 vector
+//
+// Input:
+//      - vector_length : Length of vector
+//      - in_vector     : Pointer to the vector that should be bit shifted
+//      - right_shifts  : Number of right bit shifts (negative value gives left
+//                        shifts)
+//
+// Output:
+//      - out_vector    : Pointer to the result vector (can be the same as
+//                        |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleVector(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = (gain*in_vector[k])>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Scaling gain
+//      - vector_length : Elements in the |in_vector|
+//      - right_shifts  : Number of right bit shifts applied
+//
+// Output:
+//      - out_vector    : Output vector (can be the same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleVectorWithSat(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = SATURATE( (gain*in_vector[k])>>right_shifts )
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Scaling gain
+//      - vector_length : Elements in the |in_vector|
+//      - right_shifts  : Number of right bit shifts applied
+//
+// Output:
+//      - out_vector    : Output vector (can be the same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleAndAddVectors(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = (gain1*in_vector1[k])>>right_shifts1
+//                  + (gain2*in_vector2[k])>>right_shifts2
+//
+// Input:
+//      - in_vector1    : Input vector 1
+//      - gain1         : Gain to be used for vector 1
+//      - right_shifts1 : Right bit shift to be used for vector 1
+//      - in_vector2    : Input vector 2
+//      - gain2         : Gain to be used for vector 2
+//      - right_shifts2 : Right bit shift to be used for vector 2
+//      - vector_length : Elements in the input vectors
+//
+// Output:
+//      - out_vector    : Output vector
+//
+
+//
+// WebRtcSpl_ScaleAndAddVectorsWithRound(...)
+//
+// Performs the vector operation:
+//
+//  out_vector[k] = ((scale1*in_vector1[k]) + (scale2*in_vector2[k])
+//                      + round_value) >> right_shifts
+//
+//      where:
+//
+//  round_value = (1<<right_shifts)>>1
+//
+// Input:
+//      - in_vector1    : Input vector 1
+//      - scale1        : Gain to be used for vector 1
+//      - in_vector2    : Input vector 2
+//      - scale2        : Gain to be used for vector 2
+//      - right_shifts  : Number of right bit shifts to be applied
+//      - vector_length : Number of elements in the input vectors
+//
+// Output:
+//      - out_vector    : Output vector
+//
+
+//
+// WebRtcSpl_ReverseOrderMultArrayElements(...)
+//
+// Performs the vector operation:
+//  out_vector[n] = (in_vector[n]*window[-n])>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - window        : Window vector (should be reversed). The pointer
+//                        should be set to the last value in the vector
+//      - right_shifts  : Number of right bit shift to be applied after the
+//                        multiplication
+//      - vector_length : Number of elements in |in_vector|
+//
+// Output:
+//      - out_vector    : Output vector (can be same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ElementwiseVectorMult(...)
+//
+// Performs the vector operation:
+//  out_vector[n] = (in_vector[n]*window[n])>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - window        : Window vector.
+//      - right_shifts  : Number of right bit shift to be applied after the
+//                        multiplication
+//      - vector_length : Number of elements in |in_vector|
+//
+// Output:
+//      - out_vector    : Output vector (can be same as |in_vector|)
+//
+
+//
+// WebRtcSpl_AddVectorsAndShift(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = (in_vector1[k] + in_vector2[k])>>right_shifts
+//
+// Input:
+//      - in_vector1    : Input vector 1
+//      - in_vector2    : Input vector 2
+//      - right_shifts  : Number of right bit shift to be applied after the
+//                        multiplication
+//      - vector_length : Number of elements in |in_vector1| and |in_vector2|
+//
+// Output:
+//      - out_vector    : Output vector (can be same as |in_vector1|)
+//
+
+//
+// WebRtcSpl_AddAffineVectorToVector(...)
+//
+// Adds an affine transformed vector to another vector |out_vector|, i.e,
+// performs
+//  out_vector[k] += (in_vector[k]*gain+add_constant)>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Gain value, used to multiply the in vector with
+//      - add_constant  : Constant value to add (usually 1<<(right_shifts-1),
+//                        but others can be used as well
+//      - right_shifts  : Number of right bit shifts (0-16)
+//      - vector_length : Number of samples in |in_vector| and |out_vector|
+//
+// Output:
+//      - out_vector    : Vector with the output
+//
+
+//
+// WebRtcSpl_AffineTransformVector(...)
+//
+// Affine transforms a vector, i.e, performs
+//  out_vector[k] = (in_vector[k]*gain+add_constant)>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Gain value, used to multiply the in vector with
+//      - add_constant  : Constant value to add (usually 1<<(right_shifts-1),
+//                        but others can be used as well
+//      - right_shifts  : Number of right bit shifts (0-16)
+//      - vector_length : Number of samples in |in_vector| and |out_vector|
+//
+// Output:
+//      - out_vector    : Vector with the output
+//
+
+//
+// WebRtcSpl_AutoCorrelation(...)
+//
+// A 32-bit fix-point implementation of auto-correlation computation
+//
+// Input:
+//      - vector        : Vector to calculate autocorrelation upon
+//      - vector_length : Length (in samples) of |vector|
+//      - order         : The order up to which the autocorrelation should be
+//                        calculated
+//
+// Output:
+//      - result_vector : auto-correlation values (values should be seen
+//                        relative to each other since the absolute values
+//                        might have been down shifted to avoid overflow)
+//
+//      - scale         : The number of left shifts required to obtain the
+//                        auto-correlation in Q0
+//
+// Return value         : Number of samples in |result_vector|, i.e., (order+1)
+//
+
+//
+// WebRtcSpl_LevinsonDurbin(...)
+//
+// A 32-bit fix-point implementation of the Levinson-Durbin algorithm that
+// does NOT use the 64 bit class
+//
+// Input:
+//      - auto_corr : Vector with autocorrelation values of length >=
+//                    |use_order|+1
+//      - use_order : The LPC filter order (support up to order 20)
+//
+// Output:
+//      - lpc_coef  : lpc_coef[0..use_order] LPC coefficients in Q12
+//      - refl_coef : refl_coef[0...use_order-1]| Reflection coefficients in
+//                    Q15
+//
+// Return value     : 1 for stable 0 for unstable
+//
+
+//
+// WebRtcSpl_ReflCoefToLpc(...)
+//
+// Converts reflection coefficients |refl_coef| to LPC coefficients |lpc_coef|.
+// This version is a 16 bit operation.
+//
+// NOTE: The 16 bit refl_coef -> lpc_coef conversion might result in a
+// "slightly unstable" filter (i.e., a pole just outside the unit circle) in
+// "rare" cases even if the reflection coefficients are stable.
+//
+// Input:
+//      - refl_coef : Reflection coefficients in Q15 that should be converted
+//                    to LPC coefficients
+//      - use_order : Number of coefficients in |refl_coef|
+//
+// Output:
+//      - lpc_coef  : LPC coefficients in Q12
+//
+
+//
+// WebRtcSpl_LpcToReflCoef(...)
+//
+// Converts LPC coefficients |lpc_coef| to reflection coefficients |refl_coef|.
+// This version is a 16 bit operation.
+// The conversion is implemented by the step-down algorithm.
+//
+// Input:
+//      - lpc_coef  : LPC coefficients in Q12, that should be converted to
+//                    reflection coefficients
+//      - use_order : Number of coefficients in |lpc_coef|
+//
+// Output:
+//      - refl_coef : Reflection coefficients in Q15.
+//
+
+//
+// WebRtcSpl_AutoCorrToReflCoef(...)
+//
+// Calculates reflection coefficients (16 bit) from auto-correlation values
+//
+// Input:
+//      - auto_corr : Auto-correlation values
+//      - use_order : Number of coefficients wanted be calculated
+//
+// Output:
+//      - refl_coef : Reflection coefficients in Q15.
+//
+
+//
+// WebRtcSpl_CrossCorrelation(...)
+//
+// Calculates the cross-correlation between two sequences |vector1| and
+// |vector2|. |vector1| is fixed and |vector2| slides as the pointer is
+// increased with the amount |step_vector2|
+//
+// Input:
+//      - vector1           : First sequence (fixed throughout the correlation)
+//      - vector2           : Second sequence (slides |step_vector2| for each
+//                            new correlation)
+//      - dim_vector        : Number of samples to use in the cross-correlation
+//      - dim_cross_corr    : Number of cross-correlations to calculate (the
+//                            start position for |vector2| is updated for each
+//                            new one)
+//      - right_shifts      : Number of right bit shifts to use. This will
+//                            become the output Q-domain.
+//      - step_vector2      : How many (positive or negative) steps the
+//                            |vector2| pointer should be updated for each new
+//                            cross-correlation value.
+//
+// Output:
+//      - cross_corr        : The cross-correlation in Q(-right_shifts)
+//
+
+//
+// WebRtcSpl_GetHanningWindow(...)
+//
+// Creates (the first half of) a Hanning window. Size must be at least 1 and
+// at most 512.
+//
+// Input:
+//      - size      : Length of the requested Hanning window (1 to 512)
+//
+// Output:
+//      - window    : Hanning vector in Q14.
+//
+
+//
+// WebRtcSpl_SqrtOfOneMinusXSquared(...)
+//
+// Calculates y[k] = sqrt(1 - x[k]^2) for each element of the input vector
+// |in_vector|. Input and output values are in Q15.
+//
+// Inputs:
+//      - in_vector     : Values to calculate sqrt(1 - x^2) of
+//      - vector_length : Length of vector |in_vector|
+//
+// Output:
+//      - out_vector    : Output values in Q15
+//
+
+//
+// WebRtcSpl_IncreaseSeed(...)
+//
+// Increases the seed (and returns the new value)
+//
+// Input:
+//      - seed      : Seed for random calculation
+//
+// Output:
+//      - seed      : Updated seed value
+//
+// Return value     : The new seed value
+//
+
+//
+// WebRtcSpl_RandU(...)
+//
+// Produces a uniformly distributed value in the WebRtc_Word16 range
+//
+// Input:
+//      - seed      : Seed for random calculation
+//
+// Output:
+//      - seed      : Updated seed value
+//
+// Return value     : Uniformly distributed value in the range
+//                    [Word16_MIN...Word16_MAX]
+//
+
+//
+// WebRtcSpl_RandN(...)
+//
+// Produces a normal distributed value in the WebRtc_Word16 range
+//
+// Input:
+//      - seed      : Seed for random calculation
+//
+// Output:
+//      - seed      : Updated seed value
+//
+// Return value     : N(0,1) value in the Q13 domain
+//
+
+//
+// WebRtcSpl_RandUArray(...)
+//
+// Produces a uniformly distributed vector with elements in the WebRtc_Word16
+// range
+//
+// Input:
+//      - vector_length : Samples wanted in the vector
+//      - seed          : Seed for random calculation
+//
+// Output:
+//      - vector        : Vector with the uniform values
+//      - seed          : Updated seed value
+//
+// Return value         : Number of samples in vector, i.e., |vector_length|
+//
+
+//
+// WebRtcSpl_Sqrt(...)
+//
+// Returns the square root of the input value |value|. The precision of this
+// function is integer precision, i.e., sqrt(8) gives 2 as answer.
+// If |value| is a negative number then 0 is returned.
+//
+// Algorithm:
+//
+// A sixth order Taylor Series expansion is used here to compute the square
+// root of a number y^0.5 = (1+x)^0.5
+// where
+// x = y-1
+//   = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
+// 0.5 <= x < 1
+//
+// Input:
+//      - value     : Value to calculate sqrt of
+//
+// Return value     : Result of the sqrt calculation
+//
+
+//
+// WebRtcSpl_SqrtFloor(...)
+//
+// Returns the square root of the input value |value|. The precision of this
+// function is rounding down integer precision, i.e., sqrt(8) gives 2 as answer.
+// If |value| is a negative number then 0 is returned.
+//
+// Algorithm:
+//
+// An iterative 4 cylce/bit routine
+//
+// Input:
+//      - value     : Value to calculate sqrt of
+//
+// Return value     : Result of the sqrt calculation
+//
+
+//
+// WebRtcSpl_DivU32U16(...)
+//
+// Divides a WebRtc_UWord32 |num| by a WebRtc_UWord16 |den|.
+//
+// If |den|==0, (WebRtc_UWord32)0xFFFFFFFF is returned.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division (as a WebRtc_UWord32), i.e., the
+//                    integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivW32W16(...)
+//
+// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|.
+//
+// If |den|==0, (WebRtc_Word32)0x7FFFFFFF is returned.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division (as a WebRtc_Word32), i.e., the
+//                    integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivW32W16ResW16(...)
+//
+// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|, assuming that the
+// result is less than 32768, otherwise an unpredictable result will occur.
+//
+// If |den|==0, (WebRtc_Word16)0x7FFF is returned.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division (as a WebRtc_Word16), i.e., the
+//                    integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivResultInQ31(...)
+//
+// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|, assuming that the
+// absolute value of the denominator is larger than the numerator, otherwise
+// an unpredictable result will occur.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division in Q31.
+//
+
+//
+// WebRtcSpl_DivW32HiLow(...)
+//
+// Divides a WebRtc_Word32 |num| by a denominator in hi, low format. The
+// absolute value of the denominator has to be larger (or equal to) the
+// numerator.
+//
+// Input:
+//      - num       : Numerator
+//      - den_hi    : High part of denominator
+//      - den_low   : Low part of denominator
+//
+// Return value     : Divided value in Q31
+//
+
+//
+// WebRtcSpl_Energy(...)
+//
+// Calculates the energy of a vector
+//
+// Input:
+//      - vector        : Vector which the energy should be calculated on
+//      - vector_length : Number of samples in vector
+//
+// Output:
+//      - scale_factor  : Number of left bit shifts needed to get the physical
+//                        energy value, i.e, to get the Q0 value
+//
+// Return value         : Energy value in Q(-|scale_factor|)
+//
+
+//
+// WebRtcSpl_FilterAR(...)
+//
+// Performs a 32-bit AR filtering on a vector in Q12
+//
+// Input:
+//  - ar_coef                   : AR-coefficient vector (values in Q12),
+//                                ar_coef[0] must be 4096.
+//  - ar_coef_length            : Number of coefficients in |ar_coef|.
+//  - in_vector                 : Vector to be filtered.
+//  - in_vector_length          : Number of samples in |in_vector|.
+//  - filter_state              : Current state (higher part) of the filter.
+//  - filter_state_length       : Length (in samples) of |filter_state|.
+//  - filter_state_low          : Current state (lower part) of the filter.
+//  - filter_state_low_length   : Length (in samples) of |filter_state_low|.
+//  - out_vector_low_length     : Maximum length (in samples) of
+//                                |out_vector_low|.
+//
+// Output:
+//  - filter_state              : Updated state (upper part) vector.
+//  - filter_state_low          : Updated state (lower part) vector.
+//  - out_vector                : Vector containing the upper part of the
+//                                filtered values.
+//  - out_vector_low            : Vector containing the lower part of the
+//                                filtered values.
+//
+// Return value                 : Number of samples in the |out_vector|.
+//
+
+//
+// WebRtcSpl_FilterMAFastQ12(...)
+//
+// Performs a MA filtering on a vector in Q12
+//
+// Input:
+//      - in_vector         : Input samples (state in positions
+//                            in_vector[-order] .. in_vector[-1])
+//      - ma_coef           : Filter coefficients (in Q12)
+//      - ma_coef_length    : Number of B coefficients (order+1)
+//      - vector_length     : Number of samples to be filtered
+//
+// Output:
+//      - out_vector        : Filtered samples
+//
+
+//
+// WebRtcSpl_FilterARFastQ12(...)
+//
+// Performs a AR filtering on a vector in Q12
+//
+// Input:
+//      - in_vector         : Input samples
+//      - out_vector        : State information in positions
+//                            out_vector[-order] .. out_vector[-1]
+//      - ar_coef           : Filter coefficients (in Q12)
+//      - ar_coef_length    : Number of B coefficients (order+1)
+//      - vector_length     : Number of samples to be filtered
+//
+// Output:
+//      - out_vector        : Filtered samples
+//
+
+//
+// WebRtcSpl_DownsampleFast(...)
+//
+// Performs a MA down sampling filter on a vector
+//
+// Input:
+//      - in_vector         : Input samples (state in positions
+//                            in_vector[-order] .. in_vector[-1])
+//      - in_vector_length  : Number of samples in |in_vector| to be filtered.
+//                            This must be at least
+//                            |delay| + |factor|*(|out_vector_length|-1) + 1)
+//      - out_vector_length : Number of down sampled samples desired
+//      - ma_coef           : Filter coefficients (in Q12)
+//      - ma_coef_length    : Number of B coefficients (order+1)
+//      - factor            : Decimation factor
+//      - delay             : Delay of filter (compensated for in out_vector)
+//
+// Output:
+//      - out_vector        : Filtered samples
+//
+// Return value             : 0 if OK, -1 if |in_vector| is too short
+//
+
+//
+// WebRtcSpl_DotProductWithScale(...)
+//
+// Calculates the dot product between two (WebRtc_Word16) vectors
+//
+// Input:
+//      - vector1       : Vector 1
+//      - vector2       : Vector 2
+//      - vector_length : Number of samples used in the dot product
+//      - scaling       : The number of right bit shifts to apply on each term
+//                        during calculation to avoid overflow, i.e., the
+//                        output will be in Q(-|scaling|)
+//
+// Return value         : The dot product in Q(-scaling)
+//
+
+//
+// WebRtcSpl_ComplexIFFT(...)
+//
+// Complex Inverse FFT
+//
+// Computes an inverse complex 2^|stages|-point FFT on the input vector, which
+// is in bit-reversed order. The original content of the vector is destroyed in
+// the process, since the input is overwritten by the output, normal-ordered,
+// FFT vector. With X as the input complex vector, y as the output complex
+// vector and with M = 2^|stages|, the following is computed:
+//
+//        M-1
+// y(k) = sum[X(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
+//        i=0
+//
+// The implementations are optimized for speed, not for code size. It uses the
+// decimation-in-time algorithm with radix-2 butterfly technique.
+//
+// Input:
+//      - vector    : In pointer to complex vector containing 2^|stages|
+//                    real elements interleaved with 2^|stages| imaginary
+//                    elements.
+//                    [ReImReImReIm....]
+//                    The elements are in Q(-scale) domain, see more on Return
+//                    Value below.
+//
+//      - stages    : Number of FFT stages. Must be at least 3 and at most 10,
+//                    since the table WebRtcSpl_kSinTable1024[] is 1024
+//                    elements long.
+//
+//      - mode      : This parameter gives the user to choose how the FFT
+//                    should work.
+//                    mode==0: Low-complexity and Low-accuracy mode
+//                    mode==1: High-complexity and High-accuracy mode
+//
+// Output:
+//      - vector    : Out pointer to the FFT vector (the same as input).
+//
+// Return Value     : The scale value that tells the number of left bit shifts
+//                    that the elements in the |vector| should be shifted with
+//                    in order to get Q0 values, i.e. the physically correct
+//                    values. The scale parameter is always 0 or positive,
+//                    except if N>1024 (|stages|>10), which returns a scale
+//                    value of -1, indicating error.
+//
+
+//
+// WebRtcSpl_ComplexFFT(...)
+//
+// Complex FFT
+//
+// Computes a complex 2^|stages|-point FFT on the input vector, which is in
+// bit-reversed order. The original content of the vector is destroyed in
+// the process, since the input is overwritten by the output, normal-ordered,
+// FFT vector. With x as the input complex vector, Y as the output complex
+// vector and with M = 2^|stages|, the following is computed:
+//
+//              M-1
+// Y(k) = 1/M * sum[x(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
+//              i=0
+//
+// The implementations are optimized for speed, not for code size. It uses the
+// decimation-in-time algorithm with radix-2 butterfly technique.
+//
+// This routine prevents overflow by scaling by 2 before each FFT stage. This is
+// a fixed scaling, for proper normalization - there will be log2(n) passes, so
+// this results in an overall factor of 1/n, distributed to maximize arithmetic
+// accuracy.
+//
+// Input:
+//      - vector    : In pointer to complex vector containing 2^|stages| real
+//                    elements interleaved with 2^|stages| imaginary elements.
+//                    [ReImReImReIm....]
+//                    The output is in the Q0 domain.
+//
+//      - stages    : Number of FFT stages. Must be at least 3 and at most 10,
+//                    since the table WebRtcSpl_kSinTable1024[] is 1024
+//                    elements long.
+//
+//      - mode      : This parameter gives the user to choose how the FFT
+//                    should work.
+//                    mode==0: Low-complexity and Low-accuracy mode
+//                    mode==1: High-complexity and High-accuracy mode
+//
+// Output:
+//      - vector    : The output FFT vector is in the Q0 domain.
+//
+// Return value     : The scale parameter is always 0, except if N>1024,
+//                    which returns a scale value of -1, indicating error.
+//
+
+//
+// WebRtcSpl_ComplexBitReverse(...)
+//
+// Complex Bit Reverse
+//
+// This function bit-reverses the position of elements in the complex input
+// vector into the output vector.
+//
+// If you bit-reverse a linear-order array, you obtain a bit-reversed order
+// array. If you bit-reverse a bit-reversed order array, you obtain a
+// linear-order array.
+//
+// Input:
+//      - vector    : In pointer to complex vector containing 2^|stages| real
+//                    elements interleaved with 2^|stages| imaginary elements.
+//                    [ReImReImReIm....]
+//      - stages    : Number of FFT stages. Must be at least 3 and at most 10,
+//                    since the table WebRtcSpl_kSinTable1024[] is 1024
+//                    elements long.
+//
+// Output:
+//      - vector    : Out pointer to complex vector in bit-reversed order.
+//                    The input vector is over written.
+//
+
+//
+// WebRtcSpl_AnalysisQMF(...)
+//
+// Splits a 0-2*F Hz signal into two sub bands: 0-F Hz and F-2*F Hz. The
+// current version has F = 8000, therefore, a super-wideband audio signal is
+// split to lower-band 0-8 kHz and upper-band 8-16 kHz.
+//
+// Input:
+//      - in_data       : Wide band speech signal, 320 samples (10 ms)
+//
+// Input & Output:
+//      - filter_state1 : Filter state for first All-pass filter
+//      - filter_state2 : Filter state for second All-pass filter
+//
+// Output:
+//      - low_band      : Lower-band signal 0-8 kHz band, 160 samples (10 ms)
+//      - high_band     : Upper-band signal 8-16 kHz band (flipped in frequency
+//                        domain), 160 samples (10 ms)
+//
+
+//
+// WebRtcSpl_SynthesisQMF(...)
+//
+// Combines the two sub bands (0-F and F-2*F Hz) into a signal of 0-2*F
+// Hz, (current version has F = 8000 Hz). So the filter combines lower-band
+// (0-8 kHz) and upper-band (8-16 kHz) channels to obtain super-wideband 0-16
+// kHz audio.
+//
+// Input:
+//      - low_band      : The signal with the 0-8 kHz band, 160 samples (10 ms)
+//      - high_band     : The signal with the 8-16 kHz band, 160 samples (10 ms)
+//
+// Input & Output:
+//      - filter_state1 : Filter state for first All-pass filter
+//      - filter_state2 : Filter state for second All-pass filter
+//
+// Output:
+//      - out_data      : Super-wideband speech signal, 0-16 kHz
+//
+
+// WebRtc_Word16 WebRtcSpl_SatW32ToW16(...)
+//
+// This function saturates a 32-bit word into a 16-bit word.
+// 
+// Input:
+//      - value32   : The value of a 32-bit word.
+//
+// Output:
+//      - out16     : the saturated 16-bit word.
+//
+
+// int32_t WebRtc_MulAccumW16(...)
+//
+// This function multiply a 16-bit word by a 16-bit word, and accumulate this
+// value to a 32-bit integer.
+// 
+// Input:
+//      - a    : The value of the first 16-bit word.
+//      - b    : The value of the second 16-bit word.
+//      - c    : The value of an 32-bit integer.
+//
+// Return Value: The value of a * b + c.
+//
+
+// WebRtc_Word16 WebRtcSpl_get_version(...)
+//
+// This function gives the version string of the Signal Processing Library.
+//
+// Input:
+//      - length_in_bytes   : The size of Allocated space (in Bytes) where
+//                            the version number is written to (in string format).
+//
+// Output:
+//      - version           : Pointer to a buffer where the version number is written to.
+//
diff --git a/mozillaspeechlibrary/src/main/cpp/spl_inl.h b/mozillaspeechlibrary/src/main/cpp/spl_inl.h
new file mode 100644
index 0000000..8915bd8
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/spl_inl.h
@@ -0,0 +1,159 @@
+/*
+ *  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_
diff --git a/mozillaspeechlibrary/src/main/cpp/spl_inl_armv7.h b/mozillaspeechlibrary/src/main/cpp/spl_inl_armv7.h
new file mode 100644
index 0000000..9299cc8
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/spl_inl_armv7.h
@@ -0,0 +1,137 @@
+/*
+ *  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 %0, #16, %1" : "=r"(out16) : "r"(value32));
+
+  return out16;
+}
+#endif  // WEBRTC_SPL_SPL_INL_ARMV7_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/typedefs.h b/mozillaspeechlibrary/src/main/cpp/typedefs.h
new file mode 100644
index 0000000..56ef060
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/typedefs.h
@@ -0,0 +1,145 @@
+/*
+ *  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 platform-specific typedefs and defines.
+
+#ifndef WEBRTC_TYPEDEFS_H_
+#define WEBRTC_TYPEDEFS_H_
+
+// Reserved words definitions
+// TODO(andrew): Look at removing these.
+#define WEBRTC_EXTERN extern
+#define G_CONST const
+#define WEBRTC_INLINE extern __inline
+
+// Define WebRTC preprocessor identifiers based on the current build platform.
+// TODO(andrew): Clean these up. We can probably remove everything in this
+// block.
+//   - TARGET_MAC_INTEL and TARGET_MAC aren't used anywhere.
+//   - In the few places where TARGET_PC is used, it should be replaced by
+//     something more specific.
+//   - Do we really support PowerPC? Probably not. Remove WEBRTC_MAC_INTEL
+//     from build/common.gypi as well.
+#if defined(WIN32)
+    // Windows & Windows Mobile.
+    #if !defined(WEBRTC_TARGET_PC)
+        #define WEBRTC_TARGET_PC
+    #endif
+#elif defined(__APPLE__)
+    // Mac OS X.
+    #if defined(__LITTLE_ENDIAN__ )
+        #if !defined(WEBRTC_TARGET_MAC_INTEL)
+            #define WEBRTC_TARGET_MAC_INTEL
+        #endif
+    #else
+        #if !defined(WEBRTC_TARGET_MAC)
+            #define WEBRTC_TARGET_MAC
+        #endif
+    #endif
+#else
+    // Linux etc.
+    #if !defined(WEBRTC_TARGET_PC)
+        #define WEBRTC_TARGET_PC
+    #endif
+#endif
+
+// Derived from Chromium's build/build_config.h
+// Processor architecture detection.  For more info on what's defined, see:
+//   http://msdn.microsoft.com/en-us/library/b0084kay.aspx
+//   http://www.agner.org/optimize/calling_conventions.pdf
+//   or with gcc, run: "echo | gcc -E -dM -"
+// TODO(andrew): replace WEBRTC_LITTLE_ENDIAN with WEBRTC_ARCH_LITTLE_ENDIAN?
+#if defined(_M_X64) || defined(__x86_64__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86_64
+#define WEBRTC_ARCH_64_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(_M_IX86) || defined(__i386__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__ARMEL__)
+// TODO(andrew): We'd prefer to control platform defines here, but this is
+// currently provided by the Android makefiles. Commented to avoid duplicate
+// definition warnings.
+//#define WEBRTC_ARCH_ARM
+// TODO(andrew): Chromium uses the following two defines. Should we switch?
+//#define WEBRTC_ARCH_ARM_FAMILY
+//#define WEBRTC_ARCH_ARMEL
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__mips__)
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#else
+#error Please add support for your architecture in typedefs.h
+#endif
+
+#if defined(__SSE2__) || defined(_MSC_VER)
+#define WEBRTC_USE_SSE2
+#endif
+
+#if defined(WEBRTC_TARGET_PC)
+
+#if !defined(_MSC_VER)
+  #include <stdint.h>
+#else
+    // Define C99 equivalent types.
+    // Since MSVC doesn't include these headers, we have to write our own
+    // version to provide a compatibility layer between MSVC and the WebRTC
+    // headers.
+    typedef signed char         int8_t;
+    typedef signed short        int16_t;
+    typedef signed int          int32_t;
+    typedef signed long long    int64_t;
+    typedef unsigned char       uint8_t;
+    typedef unsigned short      uint16_t;
+    typedef unsigned int        uint32_t;
+    typedef unsigned long long  uint64_t;
+#endif
+
+#if defined(WIN32)
+    typedef __int64             WebRtc_Word64;
+    typedef unsigned __int64    WebRtc_UWord64;
+#else
+    typedef int64_t             WebRtc_Word64;
+    typedef uint64_t            WebRtc_UWord64;
+#endif
+    typedef int32_t             WebRtc_Word32;
+    typedef uint32_t            WebRtc_UWord32;
+    typedef int16_t             WebRtc_Word16;
+    typedef uint16_t            WebRtc_UWord16;
+    typedef char                WebRtc_Word8;
+    typedef uint8_t             WebRtc_UWord8;
+
+    // Define endian for the platform
+    #define WEBRTC_LITTLE_ENDIAN
+
+#elif defined(WEBRTC_TARGET_MAC_INTEL)
+    #include <stdint.h>
+
+    typedef int64_t             WebRtc_Word64;
+    typedef uint64_t            WebRtc_UWord64;
+    typedef int32_t             WebRtc_Word32;
+    typedef uint32_t            WebRtc_UWord32;
+    typedef int16_t             WebRtc_Word16;
+    typedef char                WebRtc_Word8;
+    typedef uint16_t            WebRtc_UWord16;
+    typedef uint8_t             WebRtc_UWord8;
+
+    // Define endian for the platform
+    #define WEBRTC_LITTLE_ENDIAN
+
+#else
+    #error "No platform defined for WebRTC type definitions (typedefs.h)"
+#endif
+
+#endif  // WEBRTC_TYPEDEFS_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/audio_util.cc b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/audio_util.cc
new file mode 100644
index 0000000..c4f9d81
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/audio_util.cc
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2013 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 "include/audio_util.h"
+
+#include "include/typedefs.h"
+
+namespace webrtc {
+
+void Deinterleave(const int16_t* interleaved, int samples_per_channel,
+                  int num_channels, int16_t** deinterleaved) {
+  for (int i = 0; i < num_channels; i++) {
+    int16_t* channel = deinterleaved[i];
+    int interleaved_idx = i;
+    for (int j = 0; j < samples_per_channel; j++) {
+      channel[j] = interleaved[interleaved_idx];
+      interleaved_idx += num_channels;
+    }
+  }
+}
+
+void Interleave(const int16_t* const* deinterleaved, int samples_per_channel,
+                int num_channels, int16_t* interleaved) {
+  for (int i = 0; i < num_channels; ++i) {
+    const int16_t* channel = deinterleaved[i];
+    int interleaved_idx = i;
+    for (int j = 0; j < samples_per_channel; j++) {
+      interleaved[interleaved_idx] = channel[j];
+      interleaved_idx += num_channels;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/include/audio_util.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/include/audio_util.h
new file mode 100644
index 0000000..dc9443f
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/include/audio_util.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2013 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_AUDIO_INCLUDE_AUDIO_UTIL_H_
+#define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
+
+#include "typedefs.h"
+
+namespace webrtc {
+
+// Clamp the floating |value| to the range representable by an int16_t.
+static inline float ClampInt16(float value) {
+  const float kMaxInt16 = 32767.f;
+  const float kMinInt16 = -32768.f;
+  return value < kMinInt16 ? kMinInt16 :
+      (value > kMaxInt16 ? kMaxInt16 : value);
+}
+
+// Return a rounded int16_t of the floating |value|. Doesn't handle overflow;
+// use ClampInt16 if necessary.
+static inline int16_t RoundToInt16(float value) {
+  return static_cast<int16_t>(value < 0.f ? value - 0.5f : value + 0.5f);
+}
+
+// Deinterleave audio from |interleaved| to the channel buffers pointed to
+// by |deinterleaved|. There must be sufficient space allocated in the
+// |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
+// per buffer).
+void Deinterleave(const int16_t* interleaved, int samples_per_channel,
+                  int num_channels, int16_t** deinterleaved);
+
+// Interleave audio from the channel buffers pointed to by |deinterleaved| to
+// |interleaved|. There must be sufficient space allocated in |interleaved|
+// (|samples_per_channel| * |num_channels|).
+void Interleave(const int16_t* const* deinterleaved, int samples_per_channel,
+                int num_channels, int16_t* interleaved);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/include/typedefs.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/include/typedefs.h
new file mode 100644
index 0000000..472c74b
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/include/typedefs.h
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2012 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 platform-specific typedefs and defines.
+// Much of it is derived from Chromium's build/build_config.h.
+
+#ifndef WEBRTC_TYPEDEFS_H_
+#define WEBRTC_TYPEDEFS_H_
+
+// For access to standard POSIXish features, use WEBRTC_POSIX instead of a
+// more specific macro.
+#if defined(WEBRTC_MAC) || defined(WEBRTC_LINUX) || \
+    defined(WEBRTC_ANDROID)
+#define WEBRTC_POSIX
+#endif
+
+// Processor architecture detection.  For more info on what's defined, see:
+//   http://msdn.microsoft.com/en-us/library/b0084kay.aspx
+//   http://www.agner.org/optimize/calling_conventions.pdf
+//   or with gcc, run: "echo | gcc -E -dM -"
+#if defined(_M_X64) || defined(__x86_64__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86_64
+#define WEBRTC_ARCH_64_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(_M_IX86) || defined(__i386__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__ARMEL__)
+// TODO(ajm): We'd prefer to control platform defines here, but this is
+// currently provided by the Android makefiles. Commented to avoid duplicate
+// definition warnings.
+//#define WEBRTC_ARCH_ARM
+// TODO(ajm): Chromium uses the following two defines. Should we switch?
+//#define WEBRTC_ARCH_ARM_FAMILY
+//#define WEBRTC_ARCH_ARMEL
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__MIPSEL__)
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#else
+#error Please add support for your architecture in typedefs.h
+#endif
+
+#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
+#error Define either WEBRTC_ARCH_LITTLE_ENDIAN or WEBRTC_ARCH_BIG_ENDIAN
+#endif
+
+#if defined(__SSE2__) || defined(_MSC_VER)
+#define WEBRTC_USE_SSE2
+#endif
+
+#if !defined(_MSC_VER)
+#include <stdint.h>
+#else
+// Define C99 equivalent types, since MSVC doesn't provide stdint.h.
+typedef signed char         int8_t;
+typedef signed short        int16_t;
+typedef signed int          int32_t;
+typedef __int64             int64_t;
+typedef unsigned char       uint8_t;
+typedef unsigned short      uint16_t;
+typedef unsigned int        uint32_t;
+typedef unsigned __int64    uint64_t;
+#endif
+
+// Borrowed from Chromium's base/compiler_specific.h.
+// Annotate a virtual method indicating it must be overriding a virtual
+// method in the parent class.
+// Use like:
+//   virtual void foo() OVERRIDE;
+#if defined(_MSC_VER)
+#define OVERRIDE override
+#elif defined(__clang__)
+// Clang defaults to C++03 and warns about using override. Squelch that.
+// Intentionally no push/pop here so all users of OVERRIDE ignore the warning
+// too. This is like passing -Wno-c++11-extensions, except that GCC won't die
+// (because it won't see this pragma).
+#pragma clang diagnostic ignored "-Wc++11-extensions"
+#define OVERRIDE override
+#elif defined(__GNUC__) && __cplusplus >= 201103 && \
+    (__GNUC__ * 10000 + __GNUC_MINOR__ * 100) >= 40700
+// GCC 4.7 supports explicit virtual overrides when C++11 support is enabled.
+#define OVERRIDE override
+#else
+#define OVERRIDE
+#endif
+
+// Annotate a function indicating the caller must examine the return value.
+// Use like:
+//   int foo() WARN_UNUSED_RESULT;
+// TODO(ajm): Hack to avoid multiple definitions until the base/ of webrtc and
+// libjingle are merged.
+#if !defined(WARN_UNUSED_RESULT)
+#if defined(__GNUC__)
+#define WARN_UNUSED_RESULT __attribute__((warn_unused_result))
+#else
+#define WARN_UNUSED_RESULT
+#endif
+#endif  // WARN_UNUSED_RESULT
+
+#endif  // WEBRTC_TYPEDEFS_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/Android.mk b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/Android.mk
new file mode 100644
index 0000000..aed7e73
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/Android.mk
@@ -0,0 +1,126 @@
+# Copyright (c) 2012 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.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_spl
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+    auto_corr_to_refl_coef.c \
+    auto_correlation.c \
+    complex_fft.c \
+    copy_set_operations.c \
+    cross_correlation.c \
+    division_operations.c \
+    dot_product_with_scale.c \
+    downsample_fast.c \
+    energy.c \
+    filter_ar.c \
+    filter_ma_fast_q12.c \
+    get_hanning_window.c \
+    get_scaling_square.c \
+    ilbc_specific_functions.c \
+    levinson_durbin.c \
+    lpc_to_refl_coef.c \
+    min_max_operations.c \
+    randomization_functions.c \
+    real_fft.c \
+    refl_coef_to_lpc.c \
+    resample.c \
+    resample_48khz.c \
+    resample_by_2.c \
+    resample_by_2_internal.c \
+    resample_fractional.c \
+    spl_init.c \
+    spl_sqrt.c \
+    spl_version.c \
+    splitting_filter.c \
+    sqrt_of_one_minus_x_squared.c \
+    vector_scaling_operations.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+    $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+    $(LOCAL_PATH)/include \
+    $(LOCAL_PATH)/../.. \
+    external/webrtc
+
+ifeq ($(ARCH_ARM_HAVE_ARMV7A),true)
+LOCAL_SRC_FILES += \
+    filter_ar_fast_q12_armv7.S
+else
+LOCAL_SRC_FILES += \
+    filter_ar_fast_q12.c
+endif
+
+ifeq ($(TARGET_ARCH),arm)
+LOCAL_SRC_FILES += \
+    complex_bit_reverse_arm.S \
+    spl_sqrt_floor_arm.S
+else
+LOCAL_SRC_FILES += \
+    complex_bit_reverse.c \
+    spl_sqrt_floor.c
+endif
+
+LOCAL_SHARED_LIBRARIES := libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+#########################
+# Build the neon library.
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_spl_neon
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+    cross_correlation_neon.S \
+    downsample_fast_neon.S \
+    min_max_operations_neon.S \
+    vector_scaling_operations_neon.S
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+    $(MY_WEBRTC_COMMON_DEFS) \
+    $(MY_ARM_CFLAGS_NEON)
+
+LOCAL_C_INCLUDES := \
+    $(LOCAL_PATH)/include \
+    $(LOCAL_PATH)/../.. \
+    external/webrtc
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
+endif # ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_bit_reverse.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_bit_reverse.c
new file mode 100644
index 0000000..3b10ed4
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_bit_reverse.c
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2012 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 "include/signal_processing_library.h"
+
+/* Tables for data buffer indexes that are bit reversed and thus need to be
+ * swapped. Note that, index_7[{0, 2, 4, ...}] are for the left side of the swap
+ * operations, while index_7[{1, 3, 5, ...}] are for the right side of the
+ * operation. Same for index_8.
+ */
+
+/* Indexes for the case of stages == 7. */
+static const int16_t index_7[112] = {
+  1, 64, 2, 32, 3, 96, 4, 16, 5, 80, 6, 48, 7, 112, 9, 72, 10, 40, 11, 104,
+  12, 24, 13, 88, 14, 56, 15, 120, 17, 68, 18, 36, 19, 100, 21, 84, 22, 52,
+  23, 116, 25, 76, 26, 44, 27, 108, 29, 92, 30, 60, 31, 124, 33, 66, 35, 98,
+  37, 82, 38, 50, 39, 114, 41, 74, 43, 106, 45, 90, 46, 58, 47, 122, 49, 70,
+  51, 102, 53, 86, 55, 118, 57, 78, 59, 110, 61, 94, 63, 126, 67, 97, 69,
+  81, 71, 113, 75, 105, 77, 89, 79, 121, 83, 101, 87, 117, 91, 109, 95, 125,
+  103, 115, 111, 123
+};
+
+/* Indexes for the case of stages == 8. */
+static const int16_t index_8[240] = {
+  1, 128, 2, 64, 3, 192, 4, 32, 5, 160, 6, 96, 7, 224, 8, 16, 9, 144, 10, 80,
+  11, 208, 12, 48, 13, 176, 14, 112, 15, 240, 17, 136, 18, 72, 19, 200, 20,
+  40, 21, 168, 22, 104, 23, 232, 25, 152, 26, 88, 27, 216, 28, 56, 29, 184,
+  30, 120, 31, 248, 33, 132, 34, 68, 35, 196, 37, 164, 38, 100, 39, 228, 41,
+  148, 42, 84, 43, 212, 44, 52, 45, 180, 46, 116, 47, 244, 49, 140, 50, 76,
+  51, 204, 53, 172, 54, 108, 55, 236, 57, 156, 58, 92, 59, 220, 61, 188, 62,
+  124, 63, 252, 65, 130, 67, 194, 69, 162, 70, 98, 71, 226, 73, 146, 74, 82,
+  75, 210, 77, 178, 78, 114, 79, 242, 81, 138, 83, 202, 85, 170, 86, 106, 87,
+  234, 89, 154, 91, 218, 93, 186, 94, 122, 95, 250, 97, 134, 99, 198, 101,
+  166, 103, 230, 105, 150, 107, 214, 109, 182, 110, 118, 111, 246, 113, 142,
+  115, 206, 117, 174, 119, 238, 121, 158, 123, 222, 125, 190, 127, 254, 131,
+  193, 133, 161, 135, 225, 137, 145, 139, 209, 141, 177, 143, 241, 147, 201,
+  149, 169, 151, 233, 155, 217, 157, 185, 159, 249, 163, 197, 167, 229, 171,
+  213, 173, 181, 175, 245, 179, 205, 183, 237, 187, 221, 191, 253, 199, 227,
+  203, 211, 207, 243, 215, 235, 223, 251, 239, 247
+};
+
+void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages) {
+  /* For any specific value of stages, we know exactly the indexes that are
+   * bit reversed. Currently (Feb. 2012) in WebRTC the only possible values of
+   * stages are 7 and 8, so we use tables to save unnecessary iterations and
+   * calculations for these two cases.
+   */
+  if (stages == 7 || stages == 8) {
+    int m = 0;
+    int length = 112;
+    const int16_t* index = index_7;
+
+    if (stages == 8) {
+      length = 240;
+      index = index_8;
+    }
+
+    /* Decimation in time. Swap the elements with bit-reversed indexes. */
+    for (m = 0; m < length; m += 2) {
+      /* We declare a int32_t* type pointer, to load both the 16-bit real
+       * and imaginary elements from complex_data in one instruction, reducing
+       * complexity.
+       */
+      int32_t* complex_data_ptr = (int32_t*)complex_data;
+      int32_t temp = 0;
+
+      temp = complex_data_ptr[index[m]];  /* Real and imaginary */
+      complex_data_ptr[index[m]] = complex_data_ptr[index[m + 1]];
+      complex_data_ptr[index[m + 1]] = temp;
+    }
+  }
+  else {
+    int m = 0, mr = 0, l = 0;
+    int n = 1 << stages;
+    int nn = n - 1;
+
+    /* Decimation in time - re-order data */
+    for (m = 1; m <= nn; ++m) {
+      int32_t* complex_data_ptr = (int32_t*)complex_data;
+      int32_t temp = 0;
+
+      /* Find out indexes that are bit-reversed. */
+      l = n;
+      do {
+        l >>= 1;
+      } while (l > nn - mr);
+      mr = (mr & (l - 1)) + l;
+
+      if (mr <= m) {
+        continue;
+      }
+
+      /* Swap the elements with bit-reversed indexes.
+       * This is similar to the loop in the stages == 7 or 8 cases.
+       */
+      temp = complex_data_ptr[m];  /* Real and imaginary */
+      complex_data_ptr[m] = complex_data_ptr[mr];
+      complex_data_ptr[mr] = temp;
+    }
+  }
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_fft.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_fft.c
new file mode 100644
index 0000000..aae0599
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_fft.c
@@ -0,0 +1,307 @@
+/*
+ *  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 "complex_fft_tables.h"
+#include "include/signal_processing_library.h"
+
+#define CFFTSFT 14
+#define CFFTRND 1
+#define CFFTRND2 16384
+
+#define CIFFTSFT 14
+#define CIFFTRND 1
+
+
+int WebRtcSpl_ComplexFFT(int16_t frfi[], int stages, int mode)
+{
+    int i, j, l, k, istep, n, m;
+    int16_t wr, wi;
+    int32_t tr32, ti32, qr32, qi32;
+
+    /* The 1024-value is a constant given from the size of 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 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
+                 * kSinTable1024[], and should not be changed depending on the input
+                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+                 */
+                wr = kSinTable1024[j + 256];
+                wi = -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 = (int32_t)frfi[2 * i];
+                    qi32 = (int32_t)frfi[2 * i + 1];
+                    frfi[2 * j] = (int16_t)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, 1);
+                    frfi[2 * j + 1] = (int16_t)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, 1);
+                    frfi[2 * i] = (int16_t)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, 1);
+                    frfi[2 * i + 1] = (int16_t)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
+                 * kSinTable1024[], and should not be changed depending on the input
+                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+                 */
+                wr = kSinTable1024[j + 256];
+                wi = -kSinTable1024[j];
+
+#ifdef WEBRTC_ARCH_ARM_V7
+                int32_t wri = 0;
+                int32_t frfi_r = 0;
+                __asm __volatile("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
+                    "r"((int32_t)wr), "r"((int32_t)wi));
+#endif
+
+                for (i = m; i < n; i += istep)
+                {
+                    j = i + l;
+
+#ifdef WEBRTC_ARCH_ARM_V7
+                    __asm __volatile(
+                      "pkhbt %[frfi_r], %[frfi_even], %[frfi_odd], lsl #16\n\t"
+                      "smlsd %[tr32], %[wri], %[frfi_r], %[cfftrnd]\n\t"
+                      "smladx %[ti32], %[wri], %[frfi_r], %[cfftrnd]\n\t"
+                      :[frfi_r]"+r"(frfi_r),
+                       [tr32]"=r"(tr32),
+                       [ti32]"=r"(ti32)
+                      :[frfi_even]"r"((int32_t)frfi[2*j]),
+                       [frfi_odd]"r"((int32_t)frfi[2*j +1]),
+                       [wri]"r"(wri),
+                       [cfftrnd]"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 = ((int32_t)frfi[2 * i]) << CFFTSFT;
+                    qi32 = ((int32_t)frfi[2 * i + 1]) << CFFTSFT;
+
+                    frfi[2 * j] = (int16_t)WEBRTC_SPL_RSHIFT_W32(
+                            (qr32 - tr32 + CFFTRND2), 1 + CFFTSFT);
+                    frfi[2 * j + 1] = (int16_t)WEBRTC_SPL_RSHIFT_W32(
+                            (qi32 - ti32 + CFFTRND2), 1 + CFFTSFT);
+                    frfi[2 * i] = (int16_t)WEBRTC_SPL_RSHIFT_W32(
+                            (qr32 + tr32 + CFFTRND2), 1 + CFFTSFT);
+                    frfi[2 * i + 1] = (int16_t)WEBRTC_SPL_RSHIFT_W32(
+                            (qi32 + ti32 + CFFTRND2), 1 + CFFTSFT);
+                }
+            }
+
+            --k;
+            l = istep;
+        }
+    }
+    return 0;
+}
+
+int WebRtcSpl_ComplexIFFT(int16_t frfi[], int stages, int mode)
+{
+    int i, j, l, k, istep, n, m, scale, shift;
+    int16_t wr, wi;
+    int32_t tr32, ti32, qr32, qi32;
+    int32_t tmp32, round2;
+
+    /* The 1024-value is a constant given from the size of 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 kSinTable1024[]. Do not change
+         depending on the input parameter 'stages' */
+
+    while (l < n)
+    {
+        // variable scaling, depending upon data
+        shift = 0;
+        round2 = 8192;
+
+        tmp32 = (int32_t)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
+                 * kSinTable1024[], and should not be changed depending on the input
+                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+                 */
+                wr = kSinTable1024[j + 256];
+                wi = 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 = (int32_t)frfi[2 * i];
+                    qi32 = (int32_t)frfi[2 * i + 1];
+                    frfi[2 * j] = (int16_t)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, shift);
+                    frfi[2 * j + 1] = (int16_t)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, shift);
+                    frfi[2 * i] = (int16_t)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, shift);
+                    frfi[2 * i + 1] = (int16_t)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
+                 * kSinTable1024[], and should not be changed depending on the input
+                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+                 */
+                wr = kSinTable1024[j + 256];
+                wi = kSinTable1024[j];
+
+#ifdef WEBRTC_ARCH_ARM_V7
+                int32_t wri = 0;
+                int32_t frfi_r = 0;
+                __asm __volatile("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
+                    "r"((int32_t)wr), "r"((int32_t)wi));
+#endif
+
+                for (i = m; i < n; i += istep)
+                {
+                    j = i + l;
+
+#ifdef WEBRTC_ARCH_ARM_V7
+                    __asm __volatile(
+                      "pkhbt %[frfi_r], %[frfi_even], %[frfi_odd], lsl #16\n\t"
+                      "smlsd %[tr32], %[wri], %[frfi_r], %[cifftrnd]\n\t"
+                      "smladx %[ti32], %[wri], %[frfi_r], %[cifftrnd]\n\t"
+                      :[frfi_r]"+r"(frfi_r),
+                       [tr32]"=r"(tr32),
+                       [ti32]"=r"(ti32)
+                      :[frfi_even]"r"((int32_t)frfi[2*j]),
+                       [frfi_odd]"r"((int32_t)frfi[2*j +1]),
+                       [wri]"r"(wri),
+                       [cifftrnd]"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 = ((int32_t)frfi[2 * i]) << CIFFTSFT;
+                    qi32 = ((int32_t)frfi[2 * i + 1]) << CIFFTSFT;
+
+                    frfi[2 * j] = (int16_t)WEBRTC_SPL_RSHIFT_W32((qr32 - tr32+round2),
+                                                                       shift+CIFFTSFT);
+                    frfi[2 * j + 1] = (int16_t)WEBRTC_SPL_RSHIFT_W32(
+                            (qi32 - ti32 + round2), shift + CIFFTSFT);
+                    frfi[2 * i] = (int16_t)WEBRTC_SPL_RSHIFT_W32((qr32 + tr32 + round2),
+                                                                       shift + CIFFTSFT);
+                    frfi[2 * i + 1] = (int16_t)WEBRTC_SPL_RSHIFT_W32(
+                            (qi32 + ti32 + round2), shift + CIFFTSFT);
+                }
+            }
+
+        }
+        --k;
+        l = istep;
+    }
+    return scale;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_fft_tables.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_fft_tables.h
new file mode 100644
index 0000000..45b0257
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/complex_fft_tables.h
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2013 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_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
+#define WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
+
+
+#include "../include/typedefs.h"
+static const int16_t 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
+};
+
+#endif  // WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/cross_correlation.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/cross_correlation.c
new file mode 100644
index 0000000..acc83dc
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/cross_correlation.c
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2012 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 "include/signal_processing_library.h"
+
+/* C version of WebRtcSpl_CrossCorrelation() for generic platforms. */
+void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation,
+                                 const int16_t* seq1,
+                                 const int16_t* seq2,
+                                 int16_t dim_seq,
+                                 int16_t dim_cross_correlation,
+                                 int16_t right_shifts,
+                                 int16_t step_seq2) {
+  int i = 0, j = 0;
+
+  for (i = 0; i < dim_cross_correlation; i++) {
+    *cross_correlation = 0;
+    /* Unrolling doesn't seem to improve performance. */
+    for (j = 0; j < dim_seq; j++) {
+      *cross_correlation += (seq1[j] * seq2[step_seq2 * i + j]) >> right_shifts;
+    }
+    cross_correlation++;
+  }
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/division_operations.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/division_operations.c
new file mode 100644
index 0000000..a2db2e6
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/division_operations.c
@@ -0,0 +1,143 @@
+/*
+ *  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 "include/signal_processing_library.h"
+
+uint32_t WebRtcSpl_DivU32U16(uint32_t num, uint16_t den)
+{
+    // Guard against division with 0
+    if (den != 0)
+    {
+        return (uint32_t)(num / den);
+    } else
+    {
+        return (uint32_t)0xFFFFFFFF;
+    }
+}
+
+int32_t WebRtcSpl_DivW32W16(int32_t num, int16_t den)
+{
+    // Guard against division with 0
+    if (den != 0)
+    {
+        return (int32_t)(num / den);
+    } else
+    {
+        return (int32_t)0x7FFFFFFF;
+    }
+}
+
+int16_t WebRtcSpl_DivW32W16ResW16(int32_t num, int16_t den)
+{
+    // Guard against division with 0
+    if (den != 0)
+    {
+        return (int16_t)(num / den);
+    } else
+    {
+        return (int16_t)0x7FFF;
+    }
+}
+
+int32_t WebRtcSpl_DivResultInQ31(int32_t num, int32_t den)
+{
+    int32_t L_num = num;
+    int32_t L_den = den;
+    int32_t 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;
+}
+
+int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
+{
+    int16_t approx, tmp_hi, tmp_low, num_hi, num_low;
+    int32_t tmpW32;
+
+    approx = (int16_t)WebRtcSpl_DivW32W16((int32_t)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 = (int32_t)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx))
+
+    // Store tmpW32 in hi and low format
+    tmp_hi = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
+    tmp_low = (int16_t)WEBRTC_SPL_RSHIFT_W32((tmpW32
+            - WEBRTC_SPL_LSHIFT_W32((int32_t)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 = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
+    tmp_low = (int16_t)WEBRTC_SPL_RSHIFT_W32((tmpW32
+            - WEBRTC_SPL_LSHIFT_W32((int32_t)tmp_hi, 16)), 1);
+
+    // Store num in hi and low format
+    num_hi = (int16_t)WEBRTC_SPL_RSHIFT_W32(num, 16);
+    num_low = (int16_t)WEBRTC_SPL_RSHIFT_W32((num
+            - WEBRTC_SPL_LSHIFT_W32((int32_t)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;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/downsample_fast.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/downsample_fast.c
new file mode 100644
index 0000000..5fad93d
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/downsample_fast.c
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2012 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 "include/signal_processing_library.h"
+
+// TODO(Bjornv): Change the function parameter order to WebRTC code style.
+// C version of WebRtcSpl_DownsampleFast() for generic platforms.
+int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
+                              int data_in_length,
+                              int16_t* data_out,
+                              int data_out_length,
+                              const int16_t* __restrict coefficients,
+                              int coefficients_length,
+                              int factor,
+                              int delay) {
+  int i = 0;
+  int j = 0;
+  int32_t out_s32 = 0;
+  int endpos = delay + factor * (data_out_length - 1) + 1;
+
+  // Return error if any of the running conditions doesn't meet.
+  if (data_out_length <= 0 || coefficients_length <= 0
+                           || data_in_length < endpos) {
+    return -1;
+  }
+
+  for (i = delay; i < endpos; i += factor) {
+    out_s32 = 2048;  // Round value, 0.5 in Q12.
+
+    for (j = 0; j < coefficients_length; j++) {
+      out_s32 += coefficients[j] * data_in[i - j];  // Q12.
+    }
+
+    out_s32 >>= 12;  // Q0.
+
+    // Saturate and store the output.
+    *data_out++ = WebRtcSpl_SatW32ToW16(out_s32);
+  }
+
+  return 0;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/energy.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/energy.c
new file mode 100644
index 0000000..bfec4d7
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/energy.c
@@ -0,0 +1,36 @@
+/*
+ *  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_Energy().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "include/signal_processing_library.h"
+
+int32_t WebRtcSpl_Energy(int16_t* vector, int vector_length, int* scale_factor)
+{
+    int32_t en = 0;
+    int i;
+    int scaling = WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length);
+    int looptimes = vector_length;
+    int16_t *vectorptr = vector;
+
+    for (i = 0; i < looptimes; i++)
+    {
+        en += WEBRTC_SPL_MUL_16_16_RSFT(*vectorptr, *vectorptr, scaling);
+        vectorptr++;
+    }
+    *scale_factor = scaling;
+
+    return en;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/get_scaling_square.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/get_scaling_square.c
new file mode 100644
index 0000000..2afa241
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/get_scaling_square.c
@@ -0,0 +1,44 @@
+/*
+ *  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_GetScalingSquare().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "include/signal_processing_library.h"
+
+int WebRtcSpl_GetScalingSquare(int16_t *in_vector, int in_vector_length, int times)
+{
+    int nbits = WebRtcSpl_GetSizeInBits(times);
+    int i;
+    int16_t smax = -1;
+    int16_t sabs;
+    int16_t *sptr = in_vector;
+    int t;
+    int looptimes = in_vector_length;
+
+    for (i = looptimes; i > 0; i--)
+    {
+        sabs = (*sptr > 0 ? *sptr++ : -*sptr++);
+        smax = (sabs > smax ? sabs : smax);
+    }
+    t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
+
+    if (smax == 0)
+    {
+        return 0; // Since norm(0) returns 0
+    } else
+    {
+        return (t > nbits) ? 0 : nbits - t;
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/real_fft.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/real_fft.h
new file mode 100644
index 0000000..983bf7a
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/real_fft.h
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2012 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_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
+#define WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
+
+#include "../../include/typedefs.h"
+
+// For ComplexFFT(), the maximum fft order is 10;
+// for OpenMax FFT in ARM, it is 12;
+// WebRTC APM uses orders of only 7 and 8.
+enum {kMaxFFTOrder = 10};
+
+struct RealFFT;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+typedef struct RealFFT* (*CreateRealFFT)(int order);
+typedef void (*FreeRealFFT)(struct RealFFT* self);
+typedef int (*RealForwardFFT)(struct RealFFT* self,
+                              const int16_t* real_data_in,
+                              int16_t* complex_data_out);
+typedef int (*RealInverseFFT)(struct RealFFT* self,
+                              const int16_t* complex_data_in,
+                              int16_t* real_data_out);
+
+extern CreateRealFFT WebRtcSpl_CreateRealFFT;
+extern FreeRealFFT WebRtcSpl_FreeRealFFT;
+extern RealForwardFFT WebRtcSpl_RealForwardFFT;
+extern RealInverseFFT WebRtcSpl_RealInverseFFT;
+
+struct RealFFT* WebRtcSpl_CreateRealFFTC(int order);
+void WebRtcSpl_FreeRealFFTC(struct RealFFT* self);
+
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+struct RealFFT* WebRtcSpl_CreateRealFFTNeon(int order);
+void WebRtcSpl_FreeRealFFTNeon(struct RealFFT* self);
+#endif
+
+// Compute an FFT for a real-valued signal of length of 2^order,
+// where 1 < order <= MAX_FFT_ORDER. Transform length is determined by the
+// specification structure, which must be initialized prior to calling the FFT
+// function with WebRtcSpl_CreateRealFFT().
+// The relationship between the input and output sequences can
+// be expressed in terms of the DFT, i.e.:
+//     x[n] = (2^(-scalefactor)/N)  . SUM[k=0,...,N-1] X[k].e^(jnk.2.pi/N)
+//     n=0,1,2,...N-1
+//     N=2^order.
+// The conjugate-symmetric output sequence is represented using a CCS vector,
+// which is of length N+2, and is organized as follows:
+//     Index:      0  1  2  3  4  5   . . .   N-2       N-1       N       N+1
+//     Component:  R0 0  R1 I1 R2 I2  . . .   R[N/2-1]  I[N/2-1]  R[N/2]  0
+// where R[n] and I[n], respectively, denote the real and imaginary components
+// for FFT bin 'n'. Bins  are numbered from 0 to N/2, where N is the FFT length.
+// Bin index 0 corresponds to the DC component, and bin index N/2 corresponds to
+// the foldover frequency.
+//
+// Input Arguments:
+//   self - pointer to preallocated and initialized FFT specification structure.
+//   real_data_in - the input signal. For an ARM Neon platform, it must be
+//                  aligned on a 32-byte boundary.
+//
+// Output Arguments:
+//   complex_data_out - the output complex signal with (2^order + 2) 16-bit
+//                      elements. For an ARM Neon platform, it must be different
+//                      from real_data_in, and aligned on a 32-byte boundary.
+//
+// Return Value:
+//   0  - FFT calculation is successful.
+//   -1 - Error with bad arguments (NULL pointers).
+int WebRtcSpl_RealForwardFFTC(struct RealFFT* self,
+                              const int16_t* real_data_in,
+                              int16_t* complex_data_out);
+
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+int WebRtcSpl_RealForwardFFTNeon(struct RealFFT* self,
+                                 const int16_t* real_data_in,
+                                 int16_t* complex_data_out);
+#endif
+
+// Compute the inverse FFT for a conjugate-symmetric input sequence of length of
+// 2^order, where 1 < order <= MAX_FFT_ORDER. Transform length is determined by
+// the specification structure, which must be initialized prior to calling the
+// FFT function with WebRtcSpl_CreateRealFFT().
+// For a transform of length M, the input sequence is represented using a packed
+// CCS vector of length M+2, which is explained in the comments for
+// WebRtcSpl_RealForwardFFTC above.
+//
+// Input Arguments:
+//   self - pointer to preallocated and initialized FFT specification structure.
+//   complex_data_in - the input complex signal with (2^order + 2) 16-bit
+//                     elements. For an ARM Neon platform, it must be aligned on
+//                     a 32-byte boundary.
+//
+// Output Arguments:
+//   real_data_out - the output real signal. For an ARM Neon platform, it must
+//                   be different to complex_data_in, and aligned on a 32-byte
+//                   boundary.
+//
+// Return Value:
+//   0 or a positive number - a value that the elements in the |real_data_out|
+//                            should be shifted left with in order to get
+//                            correct physical values.
+//   -1 - Error with bad arguments (NULL pointers).
+int WebRtcSpl_RealInverseFFTC(struct RealFFT* self,
+                              const int16_t* complex_data_in,
+                              int16_t* real_data_out);
+
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+int WebRtcSpl_RealInverseFFTNeon(struct RealFFT* self,
+                                 const int16_t* complex_data_in,
+                                 int16_t* real_data_out);
+#endif
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/signal_processing_library.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/signal_processing_library.h
new file mode 100644
index 0000000..2cfe85b
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/signal_processing_library.h
@@ -0,0 +1,1779 @@
+/*
+ *  Copyright (c) 2012 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 all of the fix point signal processing library (SPL) function
+ * descriptions and declarations.
+ * For specific function calls, see bottom of file.
+ */
+
+#ifndef WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+#define WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+
+#include <string.h>
+#include "../../include/typedefs.h"
+
+// Macros specific for the fixed point implementation
+#define WEBRTC_SPL_WORD16_MAX       32767
+#define WEBRTC_SPL_WORD16_MIN       -32768
+#define WEBRTC_SPL_WORD32_MAX       (int32_t)0x7fffffff
+#define WEBRTC_SPL_WORD32_MIN       (int32_t)0x80000000
+#define WEBRTC_SPL_MAX_LPC_ORDER    14
+#define WEBRTC_SPL_MAX_SEED_USED    0x80000000L
+#define WEBRTC_SPL_MIN(A, B)        (A < B ? A : B)  // Get min value
+#define WEBRTC_SPL_MAX(A, B)        (A > B ? A : B)  // Get max value
+// TODO(kma/bjorn): For the next two macros, investigate how to correct the code
+// for inputs of a = WEBRTC_SPL_WORD16_MIN or WEBRTC_SPL_WORD32_MIN.
+#define WEBRTC_SPL_ABS_W16(a) \
+    (((int16_t)a >= 0) ? ((int16_t)a) : -((int16_t)a))
+#define WEBRTC_SPL_ABS_W32(a) \
+    (((int32_t)a >= 0) ? ((int32_t)a) : -((int32_t)a))
+
+#ifdef WEBRTC_ARCH_LITTLE_ENDIAN
+#define WEBRTC_SPL_GET_BYTE(a, nr)  (((int8_t *)a)[nr])
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+    (((int8_t *)d_ptr)[index] = (val))
+#else
+#define WEBRTC_SPL_GET_BYTE(a, nr) \
+    ((((int16_t *)a)[nr >> 1]) >> (((nr + 1) & 0x1) * 8) & 0x00ff)
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+    ((int16_t *)d_ptr)[index >> 1] = \
+    ((((int16_t *)d_ptr)[index >> 1]) \
+    & (0x00ff << (8 * ((index) & 0x1)))) | (val << (8 * ((index + 1) & 0x1)))
+#endif
+
+#define WEBRTC_SPL_MUL(a, b) \
+    ((int32_t) ((int32_t)(a) * (int32_t)(b)))
+#define WEBRTC_SPL_UMUL(a, b) \
+    ((uint32_t) ((uint32_t)(a) * (uint32_t)(b)))
+#define WEBRTC_SPL_UMUL_RSFT16(a, b) \
+    ((uint32_t) ((uint32_t)(a) * (uint32_t)(b)) >> 16)
+#define WEBRTC_SPL_UMUL_16_16(a, b) \
+    ((uint32_t) (uint16_t)(a) * (uint16_t)(b))
+#define WEBRTC_SPL_UMUL_16_16_RSFT16(a, b) \
+    (((uint32_t) (uint16_t)(a) * (uint16_t)(b)) >> 16)
+#define WEBRTC_SPL_UMUL_32_16(a, b) \
+    ((uint32_t) ((uint32_t)(a) * (uint16_t)(b)))
+#define WEBRTC_SPL_UMUL_32_16_RSFT16(a, b) \
+    ((uint32_t) ((uint32_t)(a) * (uint16_t)(b)) >> 16)
+#define WEBRTC_SPL_MUL_16_U16(a, b) \
+    ((int32_t)(int16_t)(a) * (uint16_t)(b))
+#define WEBRTC_SPL_DIV(a, b) \
+    ((int32_t) ((int32_t)(a) / (int32_t)(b)))
+#define WEBRTC_SPL_UDIV(a, b) \
+    ((uint32_t) ((uint32_t)(a) / (uint32_t)(b)))
+
+#ifndef WEBRTC_ARCH_ARM_V7
+// For ARMv7 platforms, these are inline functions in spl_inl_armv7.h
+#ifndef MIPS32_LE
+// For MIPS platforms, these are inline functions in spl_inl_mips.h
+#define WEBRTC_SPL_MUL_16_16(a, b) \
+    ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
+#define WEBRTC_SPL_MUL_16_32_RSFT16(a, b) \
+    (WEBRTC_SPL_MUL_16_16(a, b >> 16) \
+     + ((WEBRTC_SPL_MUL_16_16(a, (b & 0xffff) >> 1) + 0x4000) >> 15))
+#define WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, b32) \
+    ((int32_t)(WEBRTC_SPL_MUL_16_32_RSFT16(a32a, b32) \
+    + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32) >> 16)))
+#define WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, b32) \
+    ((int32_t)(WEBRTC_SPL_MUL_16_32_RSFT16(( \
+    (int16_t)(a32 >> 16)), b32) + \
+    (WEBRTC_SPL_MUL_16_32_RSFT16(( \
+    (int16_t)((a32 & 0x0000FFFF) >> 1)), b32) >> 15)))
+#endif
+#endif
+
+#define WEBRTC_SPL_MUL_16_32_RSFT11(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 5) \
+    + (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x0200) >> 10))
+#define WEBRTC_SPL_MUL_16_32_RSFT14(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 2) \
+    + (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x1000) >> 13))
+#define WEBRTC_SPL_MUL_16_32_RSFT15(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 1) \
+    + (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x2000) >> 14))
+
+#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
+    (WEBRTC_SPL_MUL_16_16(a, b) >> (c))
+
+#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c) \
+    ((WEBRTC_SPL_MUL_16_16(a, b) + ((int32_t) \
+                                  (((int32_t)1) << ((c) - 1)))) >> (c))
+#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, b) + ((int32_t) (1 << 14))) >> 15)
+
+// C + the 32 most significant bits of A * B
+#define WEBRTC_SPL_SCALEDIFF32(A, B, C) \
+    (C + (B >> 16) * A + (((uint32_t)(0x0000FFFF & B) * A) >> 16))
+
+#define WEBRTC_SPL_ADD_SAT_W32(a, b)    WebRtcSpl_AddSatW32(a, b)
+#define WEBRTC_SPL_SAT(a, b, c)         (b > a ? a : b < c ? c : b)
+#define WEBRTC_SPL_MUL_32_16(a, b)      ((a) * (b))
+
+#define WEBRTC_SPL_SUB_SAT_W32(a, b)    WebRtcSpl_SubSatW32(a, b)
+#define WEBRTC_SPL_ADD_SAT_W16(a, b)    WebRtcSpl_AddSatW16(a, b)
+#define WEBRTC_SPL_SUB_SAT_W16(a, b)    WebRtcSpl_SubSatW16(a, b)
+
+// We cannot do casting here due to signed/unsigned problem
+#define WEBRTC_SPL_IS_NEG(a)            ((a) & 0x80000000)
+// Shifting with negative numbers allowed
+// Positive means left shift
+#define WEBRTC_SPL_SHIFT_W16(x, c) \
+    (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
+#define WEBRTC_SPL_SHIFT_W32(x, c) \
+    (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
+
+// Shifting with negative numbers not allowed
+// We cannot do casting here due to signed/unsigned problem
+#define WEBRTC_SPL_RSHIFT_W16(x, c)     ((x) >> (c))
+#define WEBRTC_SPL_LSHIFT_W16(x, c)     ((x) << (c))
+#define WEBRTC_SPL_RSHIFT_W32(x, c)     ((x) >> (c))
+#define WEBRTC_SPL_LSHIFT_W32(x, c)     ((x) << (c))
+
+#define WEBRTC_SPL_RSHIFT_U16(x, c)     ((uint16_t)(x) >> (c))
+#define WEBRTC_SPL_LSHIFT_U16(x, c)     ((uint16_t)(x) << (c))
+#define WEBRTC_SPL_RSHIFT_U32(x, c)     ((uint32_t)(x) >> (c))
+#define WEBRTC_SPL_LSHIFT_U32(x, c)     ((uint32_t)(x) << (c))
+
+#define WEBRTC_SPL_VNEW(t, n)           (t *) malloc (sizeof (t) * (n))
+#define WEBRTC_SPL_FREE                 free
+
+#define WEBRTC_SPL_RAND(a) \
+    ((int16_t)(WEBRTC_SPL_MUL_16_16_RSFT((a), 18816, 7) & 0x00007fff))
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define WEBRTC_SPL_MEMCPY_W8(v1, v2, length) \
+  memcpy(v1, v2, (length) * sizeof(char))
+#define WEBRTC_SPL_MEMCPY_W16(v1, v2, length) \
+  memcpy(v1, v2, (length) * sizeof(int16_t))
+
+#define WEBRTC_SPL_MEMMOVE_W16(v1, v2, length) \
+  memmove(v1, v2, (length) * sizeof(int16_t))
+
+// inline functions:
+#include "spl_inl.h"
+
+// Initialize SPL. Currently it contains only function pointer initialization.
+// If the underlying platform is known to be ARM-Neon (WEBRTC_ARCH_ARM_NEON
+// defined), the pointers will be assigned to code optimized for Neon; otherwise
+// if run-time Neon detection (WEBRTC_DETECT_ARM_NEON) is enabled, the pointers
+// will be assigned to either Neon code or generic C code; otherwise, generic C
+// code will be assigned.
+// Note that this function MUST be called in any application that uses SPL
+// functions.
+void WebRtcSpl_Init();
+
+// Get SPL Version
+int16_t WebRtcSpl_get_version(char* version, int16_t length_in_bytes);
+
+int WebRtcSpl_GetScalingSquare(int16_t* in_vector,
+                               int in_vector_length,
+                               int times);
+
+// Copy and set operations. Implementation in copy_set_operations.c.
+// Descriptions at bottom of file.
+void WebRtcSpl_MemSetW16(int16_t* vector,
+                         int16_t set_value,
+                         int vector_length);
+void WebRtcSpl_MemSetW32(int32_t* vector,
+                         int32_t set_value,
+                         int vector_length);
+void WebRtcSpl_MemCpyReversedOrder(int16_t* out_vector,
+                                   int16_t* in_vector,
+                                   int vector_length);
+int16_t WebRtcSpl_CopyFromEndW16(const int16_t* in_vector,
+                                 int16_t in_vector_length,
+                                 int16_t samples,
+                                 int16_t* out_vector);
+int16_t WebRtcSpl_ZerosArrayW16(int16_t* vector,
+                                int16_t vector_length);
+int16_t WebRtcSpl_ZerosArrayW32(int32_t* vector,
+                                int16_t vector_length);
+int16_t WebRtcSpl_OnesArrayW16(int16_t* vector,
+                               int16_t vector_length);
+int16_t WebRtcSpl_OnesArrayW32(int32_t* vector,
+                               int16_t vector_length);
+// End: Copy and set operations.
+
+
+// Minimum and maximum operation functions and their pointers.
+// Implementation in min_max_operations.c.
+
+// Returns the largest absolute value in a signed 16-bit vector.
+//
+// Input:
+//      - vector : 16-bit input vector.
+//      - length : Number of samples in vector.
+//
+// Return value  : Maximum absolute value in vector;
+//                 or -1, if (vector == NULL || length <= 0).
+typedef int16_t (*MaxAbsValueW16)(const int16_t* vector, int length);
+extern MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16;
+int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, int length);
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+int16_t WebRtcSpl_MaxAbsValueW16Neon(const int16_t* vector, int length);
+#endif
+#if defined(MIPS32_LE)
+int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, int length);
+#endif
+
+// Returns the largest absolute value in a signed 32-bit vector.
+//
+// Input:
+//      - vector : 32-bit input vector.
+//      - length : Number of samples in vector.
+//
+// Return value  : Maximum absolute value in vector;
+//                 or -1, if (vector == NULL || length <= 0).
+typedef int32_t (*MaxAbsValueW32)(const int32_t* vector, int length);
+extern MaxAbsValueW32 WebRtcSpl_MaxAbsValueW32;
+int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, int length);
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+int32_t WebRtcSpl_MaxAbsValueW32Neon(const int32_t* vector, int length);
+#endif
+#if defined(MIPS_DSP_R1_LE)
+int32_t WebRtcSpl_MaxAbsValueW32_mips(const int32_t* vector, int length);
+#endif
+
+// Returns the maximum value of a 16-bit vector.
+//
+// Input:
+//      - vector : 16-bit input vector.
+//      - length : Number of samples in vector.
+//
+// Return value  : Maximum sample value in |vector|.
+//                 If (vector == NULL || length <= 0) WEBRTC_SPL_WORD16_MIN
+//                 is returned. Note that WEBRTC_SPL_WORD16_MIN is a feasible
+//                 value and we can't catch errors purely based on it.
+typedef int16_t (*MaxValueW16)(const int16_t* vector, int length);
+extern MaxValueW16 WebRtcSpl_MaxValueW16;
+int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, int length);
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+int16_t WebRtcSpl_MaxValueW16Neon(const int16_t* vector, int length);
+#endif
+#if defined(MIPS32_LE)
+int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, int length);
+#endif
+
+// Returns the maximum value of a 32-bit vector.
+//
+// Input:
+//      - vector : 32-bit input vector.
+//      - length : Number of samples in vector.
+//
+// Return value  : Maximum sample value in |vector|.
+//                 If (vector == NULL || length <= 0) WEBRTC_SPL_WORD32_MIN
+//                 is returned. Note that WEBRTC_SPL_WORD32_MIN is a feasible
+//                 value and we can't catch errors purely based on it.
+typedef int32_t (*MaxValueW32)(const int32_t* vector, int length);
+extern MaxValueW32 WebRtcSpl_MaxValueW32;
+int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, int length);
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+int32_t WebRtcSpl_MaxValueW32Neon(const int32_t* vector, int length);
+#endif
+#if defined(MIPS32_LE)
+int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, int length);
+#endif
+
+// Returns the minimum value of a 16-bit vector.
+//
+// Input:
+//      - vector : 16-bit input vector.
+//      - length : Number of samples in vector.
+//
+// Return value  : Minimum sample value in |vector|.
+//                 If (vector == NULL || length <= 0) WEBRTC_SPL_WORD16_MAX
+//                 is returned. Note that WEBRTC_SPL_WORD16_MAX is a feasible
+//                 value and we can't catch errors purely based on it.
+typedef int16_t (*MinValueW16)(const int16_t* vector, int length);
+extern MinValueW16 WebRtcSpl_MinValueW16;
+int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, int length);
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+int16_t WebRtcSpl_MinValueW16Neon(const int16_t* vector, int length);
+#endif
+#if defined(MIPS32_LE)
+int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, int length);
+#endif
+
+// Returns the minimum value of a 32-bit vector.
+//
+// Input:
+//      - vector : 32-bit input vector.
+//      - length : Number of samples in vector.
+//
+// Return value  : Minimum sample value in |vector|.
+//                 If (vector == NULL || length <= 0) WEBRTC_SPL_WORD32_MAX
+//                 is returned. Note that WEBRTC_SPL_WORD32_MAX is a feasible
+//                 value and we can't catch errors purely based on it.
+typedef int32_t (*MinValueW32)(const int32_t* vector, int length);
+extern MinValueW32 WebRtcSpl_MinValueW32;
+int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, int length);
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+int32_t WebRtcSpl_MinValueW32Neon(const int32_t* vector, int length);
+#endif
+#if defined(MIPS32_LE)
+int32_t WebRtcSpl_MinValueW32_mips(const int32_t* vector, int length);
+#endif
+
+// Returns the vector index to the largest absolute value of a 16-bit vector.
+//
+// Input:
+//      - vector : 16-bit input vector.
+//      - length : Number of samples in vector.
+//
+// Return value  : Index to the maximum absolute value in vector, or -1,
+//                 if (vector == NULL || length <= 0).
+//                 If there are multiple equal maxima, return the index of the
+//                 first. -32768 will always have precedence over 32767 (despite
+//                 -32768 presenting an int16 absolute value of 32767);
+int WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, int length);
+
+// Returns the vector index to the maximum sample value of a 16-bit vector.
+//
+// Input:
+//      - vector : 16-bit input vector.
+//      - length : Number of samples in vector.
+//
+// Return value  : Index to the maximum value in vector (if multiple
+//                 indexes have the maximum, return the first);
+//                 or -1, if (vector == NULL || length <= 0).
+int WebRtcSpl_MaxIndexW16(const int16_t* vector, int length);
+
+// Returns the vector index to the maximum sample value of a 32-bit vector.
+//
+// Input:
+//      - vector : 32-bit input vector.
+//      - length : Number of samples in vector.
+//
+// Return value  : Index to the maximum value in vector (if multiple
+//                 indexes have the maximum, return the first);
+//                 or -1, if (vector == NULL || length <= 0).
+int WebRtcSpl_MaxIndexW32(const int32_t* vector, int length);
+
+// Returns the vector index to the minimum sample value of a 16-bit vector.
+//
+// Input:
+//      - vector : 16-bit input vector.
+//      - length : Number of samples in vector.
+//
+// Return value  : Index to the mimimum value in vector  (if multiple
+//                 indexes have the minimum, return the first);
+//                 or -1, if (vector == NULL || length <= 0).
+int WebRtcSpl_MinIndexW16(const int16_t* vector, int length);
+
+// Returns the vector index to the minimum sample value of a 32-bit vector.
+//
+// Input:
+//      - vector : 32-bit input vector.
+//      - length : Number of samples in vector.
+//
+// Return value  : Index to the mimimum value in vector  (if multiple
+//                 indexes have the minimum, return the first);
+//                 or -1, if (vector == NULL || length <= 0).
+int WebRtcSpl_MinIndexW32(const int32_t* vector, int length);
+
+// End: Minimum and maximum operations.
+
+
+// Vector scaling operations. Implementation in vector_scaling_operations.c.
+// Description at bottom of file.
+void WebRtcSpl_VectorBitShiftW16(int16_t* out_vector,
+                                 int16_t vector_length,
+                                 const int16_t* in_vector,
+                                 int16_t right_shifts);
+void WebRtcSpl_VectorBitShiftW32(int32_t* out_vector,
+                                 int16_t vector_length,
+                                 const int32_t* in_vector,
+                                 int16_t right_shifts);
+void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out_vector,
+                                      int vector_length,
+                                      const int32_t* in_vector,
+                                      int right_shifts);
+void WebRtcSpl_ScaleVector(const int16_t* in_vector,
+                           int16_t* out_vector,
+                           int16_t gain,
+                           int16_t vector_length,
+                           int16_t right_shifts);
+void WebRtcSpl_ScaleVectorWithSat(const int16_t* in_vector,
+                                  int16_t* out_vector,
+                                  int16_t gain,
+                                  int16_t vector_length,
+                                  int16_t right_shifts);
+void WebRtcSpl_ScaleAndAddVectors(const int16_t* in_vector1,
+                                  int16_t gain1, int right_shifts1,
+                                  const int16_t* in_vector2,
+                                  int16_t gain2, int right_shifts2,
+                                  int16_t* out_vector,
+                                  int vector_length);
+
+// The functions (with related pointer) perform the vector operation:
+//   out_vector[k] = ((scale1 * in_vector1[k]) + (scale2 * in_vector2[k])
+//        + round_value) >> right_shifts,
+//   where  round_value = (1 << right_shifts) >> 1.
+//
+// Input:
+//      - in_vector1       : Input vector 1
+//      - in_vector1_scale : Gain to be used for vector 1
+//      - in_vector2       : Input vector 2
+//      - in_vector2_scale : Gain to be used for vector 2
+//      - right_shifts     : Number of right bit shifts to be applied
+//      - length           : Number of elements in the input vectors
+//
+// Output:
+//      - out_vector       : Output vector
+// Return value            : 0 if OK, -1 if (in_vector1 == NULL
+//                           || in_vector2 == NULL || out_vector == NULL
+//                           || length <= 0 || right_shift < 0).
+typedef int (*ScaleAndAddVectorsWithRound)(const int16_t* in_vector1,
+                                           int16_t in_vector1_scale,
+                                           const int16_t* in_vector2,
+                                           int16_t in_vector2_scale,
+                                           int right_shifts,
+                                           int16_t* out_vector,
+                                           int length);
+extern ScaleAndAddVectorsWithRound WebRtcSpl_ScaleAndAddVectorsWithRound;
+int WebRtcSpl_ScaleAndAddVectorsWithRoundC(const int16_t* in_vector1,
+                                           int16_t in_vector1_scale,
+                                           const int16_t* in_vector2,
+                                           int16_t in_vector2_scale,
+                                           int right_shifts,
+                                           int16_t* out_vector,
+                                           int length);
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+int WebRtcSpl_ScaleAndAddVectorsWithRoundNeon(const int16_t* in_vector1,
+                                              int16_t in_vector1_scale,
+                                              const int16_t* in_vector2,
+                                              int16_t in_vector2_scale,
+                                              int right_shifts,
+                                              int16_t* out_vector,
+                                              int length);
+#endif
+#if defined(MIPS_DSP_R1_LE)
+int WebRtcSpl_ScaleAndAddVectorsWithRound_mips(const int16_t* in_vector1,
+                                               int16_t in_vector1_scale,
+                                               const int16_t* in_vector2,
+                                               int16_t in_vector2_scale,
+                                               int right_shifts,
+                                               int16_t* out_vector,
+                                               int length);
+#endif
+// End: Vector scaling operations.
+
+// iLBC specific functions. Implementations in ilbc_specific_functions.c.
+// Description at bottom of file.
+void WebRtcSpl_ReverseOrderMultArrayElements(int16_t* out_vector,
+                                             const int16_t* in_vector,
+                                             const int16_t* window,
+                                             int16_t vector_length,
+                                             int16_t right_shifts);
+void WebRtcSpl_ElementwiseVectorMult(int16_t* out_vector,
+                                     const int16_t* in_vector,
+                                     const int16_t* window,
+                                     int16_t vector_length,
+                                     int16_t right_shifts);
+void WebRtcSpl_AddVectorsAndShift(int16_t* out_vector,
+                                  const int16_t* in_vector1,
+                                  const int16_t* in_vector2,
+                                  int16_t vector_length,
+                                  int16_t right_shifts);
+void WebRtcSpl_AddAffineVectorToVector(int16_t* out_vector,
+                                       int16_t* in_vector,
+                                       int16_t gain,
+                                       int32_t add_constant,
+                                       int16_t right_shifts,
+                                       int vector_length);
+void WebRtcSpl_AffineTransformVector(int16_t* out_vector,
+                                     int16_t* in_vector,
+                                     int16_t gain,
+                                     int32_t add_constant,
+                                     int16_t right_shifts,
+                                     int vector_length);
+// End: iLBC specific functions.
+
+// Signal processing operations.
+
+// A 32-bit fix-point implementation of auto-correlation computation
+//
+// Input:
+//      - in_vector        : Vector to calculate autocorrelation upon
+//      - in_vector_length : Length (in samples) of |vector|
+//      - order            : The order up to which the autocorrelation should be
+//                           calculated
+//
+// Output:
+//      - result           : auto-correlation values (values should be seen
+//                           relative to each other since the absolute values
+//                           might have been down shifted to avoid overflow)
+//
+//      - scale            : The number of left shifts required to obtain the
+//                           auto-correlation in Q0
+//
+// Return value            :
+//      - -1, if |order| > |in_vector_length|;
+//      - Number of samples in |result|, i.e. (order+1), otherwise.
+int WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
+                              int in_vector_length,
+                              int order,
+                              int32_t* result,
+                              int* scale);
+
+// A 32-bit fix-point implementation of the Levinson-Durbin algorithm that
+// does NOT use the 64 bit class
+//
+// Input:
+//      - auto_corr : Vector with autocorrelation values of length >=
+//                    |use_order|+1
+//      - use_order : The LPC filter order (support up to order 20)
+//
+// Output:
+//      - lpc_coef  : lpc_coef[0..use_order] LPC coefficients in Q12
+//      - refl_coef : refl_coef[0...use_order-1]| Reflection coefficients in
+//                    Q15
+//
+// Return value     : 1 for stable 0 for unstable
+int16_t WebRtcSpl_LevinsonDurbin(int32_t* auto_corr,
+                                 int16_t* lpc_coef,
+                                 int16_t* refl_coef,
+                                 int16_t order);
+
+// Converts reflection coefficients |refl_coef| to LPC coefficients |lpc_coef|.
+// This version is a 16 bit operation.
+//
+// NOTE: The 16 bit refl_coef -> lpc_coef conversion might result in a
+// "slightly unstable" filter (i.e., a pole just outside the unit circle) in
+// "rare" cases even if the reflection coefficients are stable.
+//
+// Input:
+//      - refl_coef : Reflection coefficients in Q15 that should be converted
+//                    to LPC coefficients
+//      - use_order : Number of coefficients in |refl_coef|
+//
+// Output:
+//      - lpc_coef  : LPC coefficients in Q12
+void WebRtcSpl_ReflCoefToLpc(const int16_t* refl_coef,
+                             int use_order,
+                             int16_t* lpc_coef);
+
+// Converts LPC coefficients |lpc_coef| to reflection coefficients |refl_coef|.
+// This version is a 16 bit operation.
+// The conversion is implemented by the step-down algorithm.
+//
+// Input:
+//      - lpc_coef  : LPC coefficients in Q12, that should be converted to
+//                    reflection coefficients
+//      - use_order : Number of coefficients in |lpc_coef|
+//
+// Output:
+//      - refl_coef : Reflection coefficients in Q15.
+void WebRtcSpl_LpcToReflCoef(int16_t* lpc_coef,
+                             int use_order,
+                             int16_t* refl_coef);
+
+// Calculates reflection coefficients (16 bit) from auto-correlation values
+//
+// Input:
+//      - auto_corr : Auto-correlation values
+//      - use_order : Number of coefficients wanted be calculated
+//
+// Output:
+//      - refl_coef : Reflection coefficients in Q15.
+void WebRtcSpl_AutoCorrToReflCoef(const int32_t* auto_corr,
+                                  int use_order,
+                                  int16_t* refl_coef);
+
+// The functions (with related pointer) calculate the cross-correlation between
+// two sequences |seq1| and |seq2|.
+// |seq1| is fixed and |seq2| slides as the pointer is increased with the
+// amount |step_seq2|. Note the arguments should obey the relationship:
+// |dim_seq| - 1 + |step_seq2| * (|dim_cross_correlation| - 1) <
+//      buffer size of |seq2|
+//
+// Input:
+//      - seq1           : First sequence (fixed throughout the correlation)
+//      - seq2           : Second sequence (slides |step_vector2| for each
+//                            new correlation)
+//      - dim_seq        : Number of samples to use in the cross-correlation
+//      - dim_cross_correlation : Number of cross-correlations to calculate (the
+//                            start position for |vector2| is updated for each
+//                            new one)
+//      - right_shifts   : Number of right bit shifts to use. This will
+//                            become the output Q-domain.
+//      - step_seq2      : How many (positive or negative) steps the
+//                            |vector2| pointer should be updated for each new
+//                            cross-correlation value.
+//
+// Output:
+//      - cross_correlation : The cross-correlation in Q(-right_shifts)
+typedef void (*CrossCorrelation)(int32_t* cross_correlation,
+                                 const int16_t* seq1,
+                                 const int16_t* seq2,
+                                 int16_t dim_seq,
+                                 int16_t dim_cross_correlation,
+                                 int16_t right_shifts,
+                                 int16_t step_seq2);
+extern CrossCorrelation WebRtcSpl_CrossCorrelation;
+void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation,
+                                 const int16_t* seq1,
+                                 const int16_t* seq2,
+                                 int16_t dim_seq,
+                                 int16_t dim_cross_correlation,
+                                 int16_t right_shifts,
+                                 int16_t step_seq2);
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+void WebRtcSpl_CrossCorrelationNeon(int32_t* cross_correlation,
+                                    const int16_t* seq1,
+                                    const int16_t* seq2,
+                                    int16_t dim_seq,
+                                    int16_t dim_cross_correlation,
+                                    int16_t right_shifts,
+                                    int16_t step_seq2);
+#endif
+#if defined(MIPS32_LE)
+void WebRtcSpl_CrossCorrelation_mips(int32_t* cross_correlation,
+                                     const int16_t* seq1,
+                                     const int16_t* seq2,
+                                     int16_t dim_seq,
+                                     int16_t dim_cross_correlation,
+                                     int16_t right_shifts,
+                                     int16_t step_seq2);
+#endif
+
+// Creates (the first half of) a Hanning window. Size must be at least 1 and
+// at most 512.
+//
+// Input:
+//      - size      : Length of the requested Hanning window (1 to 512)
+//
+// Output:
+//      - window    : Hanning vector in Q14.
+void WebRtcSpl_GetHanningWindow(int16_t* window, int16_t size);
+
+// Calculates y[k] = sqrt(1 - x[k]^2) for each element of the input vector
+// |in_vector|. Input and output values are in Q15.
+//
+// Inputs:
+//      - in_vector     : Values to calculate sqrt(1 - x^2) of
+//      - vector_length : Length of vector |in_vector|
+//
+// Output:
+//      - out_vector    : Output values in Q15
+void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t* in_vector,
+                                      int vector_length,
+                                      int16_t* out_vector);
+// End: Signal processing operations.
+
+// Randomization functions. Implementations collected in
+// randomization_functions.c and descriptions at bottom of this file.
+uint32_t WebRtcSpl_IncreaseSeed(uint32_t* seed);
+int16_t WebRtcSpl_RandU(uint32_t* seed);
+int16_t WebRtcSpl_RandN(uint32_t* seed);
+int16_t WebRtcSpl_RandUArray(int16_t* vector,
+                             int16_t vector_length,
+                             uint32_t* seed);
+// End: Randomization functions.
+
+// Math functions
+int32_t WebRtcSpl_Sqrt(int32_t value);
+int32_t WebRtcSpl_SqrtFloor(int32_t value);
+
+// Divisions. Implementations collected in division_operations.c and
+// descriptions at bottom of this file.
+uint32_t WebRtcSpl_DivU32U16(uint32_t num, uint16_t den);
+int32_t WebRtcSpl_DivW32W16(int32_t num, int16_t den);
+int16_t WebRtcSpl_DivW32W16ResW16(int32_t num, int16_t den);
+int32_t WebRtcSpl_DivResultInQ31(int32_t num, int32_t den);
+int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low);
+// End: Divisions.
+
+int32_t WebRtcSpl_Energy(int16_t* vector, int vector_length, int* scale_factor);
+
+// Calculates the dot product between two (int16_t) vectors.
+//
+// Input:
+//      - vector1       : Vector 1
+//      - vector2       : Vector 2
+//      - vector_length : Number of samples used in the dot product
+//      - scaling       : The number of right bit shifts to apply on each term
+//                        during calculation to avoid overflow, i.e., the
+//                        output will be in Q(-|scaling|)
+//
+// Return value         : The dot product in Q(-scaling)
+int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,
+                                      const int16_t* vector2,
+                                      int length,
+                                      int scaling);
+
+// Filter operations.
+int WebRtcSpl_FilterAR(const int16_t* ar_coef,
+                       int ar_coef_length,
+                       const int16_t* in_vector,
+                       int in_vector_length,
+                       int16_t* filter_state,
+                       int filter_state_length,
+                       int16_t* filter_state_low,
+                       int filter_state_low_length,
+                       int16_t* out_vector,
+                       int16_t* out_vector_low,
+                       int out_vector_low_length);
+
+void WebRtcSpl_FilterMAFastQ12(int16_t* in_vector,
+                               int16_t* out_vector,
+                               int16_t* ma_coef,
+                               int16_t ma_coef_length,
+                               int16_t vector_length);
+
+// Performs a AR filtering on a vector in Q12
+// Input:
+//      - data_in            : Input samples
+//      - data_out           : State information in positions
+//                               data_out[-order] .. data_out[-1]
+//      - coefficients       : Filter coefficients (in Q12)
+//      - coefficients_length: Number of coefficients (order+1)
+//      - data_length        : Number of samples to be filtered
+// Output:
+//      - data_out           : Filtered samples
+void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
+                               int16_t* data_out,
+                               const int16_t* __restrict coefficients,
+                               int coefficients_length,
+                               int data_length);
+
+// The functions (with related pointer) perform a MA down sampling filter
+// on a vector.
+// Input:
+//      - data_in            : Input samples (state in positions
+//                               data_in[-order] .. data_in[-1])
+//      - data_in_length     : Number of samples in |data_in| to be filtered.
+//                               This must be at least
+//                               |delay| + |factor|*(|out_vector_length|-1) + 1)
+//      - data_out_length    : Number of down sampled samples desired
+//      - coefficients       : Filter coefficients (in Q12)
+//      - coefficients_length: Number of coefficients (order+1)
+//      - factor             : Decimation factor
+//      - delay              : Delay of filter (compensated for in out_vector)
+// Output:
+//      - data_out           : Filtered samples
+// Return value              : 0 if OK, -1 if |in_vector| is too short
+typedef int (*DownsampleFast)(const int16_t* data_in,
+                              int data_in_length,
+                              int16_t* data_out,
+                              int data_out_length,
+                              const int16_t* __restrict coefficients,
+                              int coefficients_length,
+                              int factor,
+                              int delay);
+extern DownsampleFast WebRtcSpl_DownsampleFast;
+int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
+                              int data_in_length,
+                              int16_t* data_out,
+                              int data_out_length,
+                              const int16_t* __restrict coefficients,
+                              int coefficients_length,
+                              int factor,
+                              int delay);
+#if (defined WEBRTC_DETECT_ARM_NEON) || (defined WEBRTC_ARCH_ARM_NEON)
+int WebRtcSpl_DownsampleFastNeon(const int16_t* data_in,
+                                 int data_in_length,
+                                 int16_t* data_out,
+                                 int data_out_length,
+                                 const int16_t* __restrict coefficients,
+                                 int coefficients_length,
+                                 int factor,
+                                 int delay);
+#endif
+#if defined(MIPS32_LE)
+int WebRtcSpl_DownsampleFast_mips(const int16_t* data_in,
+                                  int data_in_length,
+                                  int16_t* data_out,
+                                  int data_out_length,
+                                  const int16_t* __restrict coefficients,
+                                  int coefficients_length,
+                                  int factor,
+                                  int delay);
+#endif
+
+// End: Filter operations.
+
+// FFT operations
+
+int WebRtcSpl_ComplexFFT(int16_t vector[], int stages, int mode);
+int WebRtcSpl_ComplexIFFT(int16_t vector[], int stages, int mode);
+
+// Treat a 16-bit complex data buffer |complex_data| as an array of 32-bit
+// values, and swap elements whose indexes are bit-reverses of each other.
+//
+// Input:
+//      - complex_data  : Complex data buffer containing 2^|stages| real
+//                        elements interleaved with 2^|stages| imaginary
+//                        elements: [Re Im Re Im Re Im....]
+//      - stages        : Number of FFT stages. Must be at least 3 and at most
+//                        10, since the table WebRtcSpl_kSinTable1024[] is 1024
+//                        elements long.
+//
+// Output:
+//      - complex_data  : The complex data buffer.
+
+void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages);
+
+// End: FFT operations
+
+/************************************************************
+ *
+ * RESAMPLING FUNCTIONS AND THEIR STRUCTS ARE DEFINED BELOW
+ *
+ ************************************************************/
+
+/*******************************************************************
+ * resample.c
+ *
+ * Includes the following resampling combinations
+ * 22 kHz -> 16 kHz
+ * 16 kHz -> 22 kHz
+ * 22 kHz ->  8 kHz
+ *  8 kHz -> 22 kHz
+ *
+ ******************************************************************/
+
+// state structure for 22 -> 16 resampler
+typedef struct {
+  int32_t S_22_44[8];
+  int32_t S_44_32[8];
+  int32_t S_32_16[8];
+} WebRtcSpl_State22khzTo16khz;
+
+void WebRtcSpl_Resample22khzTo16khz(const int16_t* in,
+                                    int16_t* out,
+                                    WebRtcSpl_State22khzTo16khz* state,
+                                    int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state);
+
+// state structure for 16 -> 22 resampler
+typedef struct {
+  int32_t S_16_32[8];
+  int32_t S_32_22[8];
+} WebRtcSpl_State16khzTo22khz;
+
+void WebRtcSpl_Resample16khzTo22khz(const int16_t* in,
+                                    int16_t* out,
+                                    WebRtcSpl_State16khzTo22khz* state,
+                                    int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state);
+
+// state structure for 22 -> 8 resampler
+typedef struct {
+  int32_t S_22_22[16];
+  int32_t S_22_16[8];
+  int32_t S_16_8[8];
+} WebRtcSpl_State22khzTo8khz;
+
+void WebRtcSpl_Resample22khzTo8khz(const int16_t* in, int16_t* out,
+                                   WebRtcSpl_State22khzTo8khz* state,
+                                   int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state);
+
+// state structure for 8 -> 22 resampler
+typedef struct {
+  int32_t S_8_16[8];
+  int32_t S_16_11[8];
+  int32_t S_11_22[8];
+} WebRtcSpl_State8khzTo22khz;
+
+void WebRtcSpl_Resample8khzTo22khz(const int16_t* in, int16_t* out,
+                                   WebRtcSpl_State8khzTo22khz* state,
+                                   int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state);
+
+/*******************************************************************
+ * resample_fractional.c
+ * Functions for internal use in the other resample functions
+ *
+ * Includes the following resampling combinations
+ * 48 kHz -> 32 kHz
+ * 32 kHz -> 24 kHz
+ * 44 kHz -> 32 kHz
+ *
+ ******************************************************************/
+
+void WebRtcSpl_Resample48khzTo32khz(const int32_t* In, int32_t* Out,
+                                    int32_t K);
+
+void WebRtcSpl_Resample32khzTo24khz(const int32_t* In, int32_t* Out,
+                                    int32_t K);
+
+void WebRtcSpl_Resample44khzTo32khz(const int32_t* In, int32_t* Out,
+                                    int32_t K);
+
+/*******************************************************************
+ * resample_48khz.c
+ *
+ * Includes the following resampling combinations
+ * 48 kHz -> 16 kHz
+ * 16 kHz -> 48 kHz
+ * 48 kHz ->  8 kHz
+ *  8 kHz -> 48 kHz
+ *
+ ******************************************************************/
+
+typedef struct {
+  int32_t S_48_48[16];
+  int32_t S_48_32[8];
+  int32_t S_32_16[8];
+} WebRtcSpl_State48khzTo16khz;
+
+void WebRtcSpl_Resample48khzTo16khz(const int16_t* in, int16_t* out,
+                                    WebRtcSpl_State48khzTo16khz* state,
+                                    int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state);
+
+typedef struct {
+  int32_t S_16_32[8];
+  int32_t S_32_24[8];
+  int32_t S_24_48[8];
+} WebRtcSpl_State16khzTo48khz;
+
+void WebRtcSpl_Resample16khzTo48khz(const int16_t* in, int16_t* out,
+                                    WebRtcSpl_State16khzTo48khz* state,
+                                    int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state);
+
+typedef struct {
+  int32_t S_48_24[8];
+  int32_t S_24_24[16];
+  int32_t S_24_16[8];
+  int32_t S_16_8[8];
+} WebRtcSpl_State48khzTo8khz;
+
+void WebRtcSpl_Resample48khzTo8khz(const int16_t* in, int16_t* out,
+                                   WebRtcSpl_State48khzTo8khz* state,
+                                   int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state);
+
+typedef struct {
+  int32_t S_8_16[8];
+  int32_t S_16_12[8];
+  int32_t S_12_24[8];
+  int32_t S_24_48[8];
+} WebRtcSpl_State8khzTo48khz;
+
+void WebRtcSpl_Resample8khzTo48khz(const int16_t* in, int16_t* out,
+                                   WebRtcSpl_State8khzTo48khz* state,
+                                   int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state);
+
+/*******************************************************************
+ * resample_by_2.c
+ *
+ * Includes down and up sampling by a factor of two.
+ *
+ ******************************************************************/
+
+void WebRtcSpl_DownsampleBy2(const int16_t* in, int16_t len,
+                             int16_t* out, int32_t* filtState);
+
+void WebRtcSpl_UpsampleBy2(const int16_t* in, int16_t len,
+                           int16_t* out, int32_t* filtState);
+
+/************************************************************
+ * END OF RESAMPLING FUNCTIONS
+ ************************************************************/
+void WebRtcSpl_AnalysisQMF(const int16_t* in_data,
+                           int16_t* low_band,
+                           int16_t* high_band,
+                           int32_t* filter_state1,
+                           int32_t* filter_state2);
+void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
+                            const int16_t* high_band,
+                            int16_t* out_data,
+                            int32_t* filter_state1,
+                            int32_t* filter_state2);
+
+#ifdef __cplusplus
+}
+#endif  // __cplusplus
+#endif  // WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+
+//
+// WebRtcSpl_AddSatW16(...)
+// WebRtcSpl_AddSatW32(...)
+//
+// Returns the result of a saturated 16-bit, respectively 32-bit, addition of
+// the numbers specified by the |var1| and |var2| parameters.
+//
+// Input:
+//      - var1      : Input variable 1
+//      - var2      : Input variable 2
+//
+// Return value     : Added and saturated value
+//
+
+//
+// WebRtcSpl_SubSatW16(...)
+// WebRtcSpl_SubSatW32(...)
+//
+// Returns the result of a saturated 16-bit, respectively 32-bit, subtraction
+// of the numbers specified by the |var1| and |var2| parameters.
+//
+// Input:
+//      - var1      : Input variable 1
+//      - var2      : Input variable 2
+//
+// Returned value   : Subtracted and saturated value
+//
+
+//
+// WebRtcSpl_GetSizeInBits(...)
+//
+// Returns the # of bits that are needed at the most to represent the number
+// specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bits needed to represent |value|
+//
+
+//
+// WebRtcSpl_NormW32(...)
+//
+// Norm returns the # of left shifts required to 32-bit normalize the 32-bit
+// signed number specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_NormW16(...)
+//
+// Norm returns the # of left shifts required to 16-bit normalize the 16-bit
+// signed number specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_NormU32(...)
+//
+// Norm returns the # of left shifts required to 32-bit normalize the unsigned
+// 32-bit number specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_GetScalingSquare(...)
+//
+// Returns the # of bits required to scale the samples specified in the
+// |in_vector| parameter so that, if the squares of the samples are added the
+// # of times specified by the |times| parameter, the 32-bit addition will not
+// overflow (result in int32_t).
+//
+// Input:
+//      - in_vector         : Input vector to check scaling on
+//      - in_vector_length  : Samples in |in_vector|
+//      - times             : Number of additions to be performed
+//
+// Return value             : Number of right bit shifts needed to avoid
+//                            overflow in the addition calculation
+//
+
+//
+// WebRtcSpl_MemSetW16(...)
+//
+// Sets all the values in the int16_t vector |vector| of length
+// |vector_length| to the specified value |set_value|
+//
+// Input:
+//      - vector        : Pointer to the int16_t vector
+//      - set_value     : Value specified
+//      - vector_length : Length of vector
+//
+
+//
+// WebRtcSpl_MemSetW32(...)
+//
+// Sets all the values in the int32_t vector |vector| of length
+// |vector_length| to the specified value |set_value|
+//
+// Input:
+//      - vector        : Pointer to the int16_t vector
+//      - set_value     : Value specified
+//      - vector_length : Length of vector
+//
+
+//
+// WebRtcSpl_MemCpyReversedOrder(...)
+//
+// Copies all the values from the source int16_t vector |in_vector| to a
+// destination int16_t vector |out_vector|. It is done in reversed order,
+// meaning that the first sample of |in_vector| is copied to the last sample of
+// the |out_vector|. The procedure continues until the last sample of
+// |in_vector| has been copied to the first sample of |out_vector|. This
+// creates a reversed vector. Used in e.g. prediction in iLBC.
+//
+// Input:
+//      - in_vector     : Pointer to the first sample in a int16_t vector
+//                        of length |length|
+//      - vector_length : Number of elements to copy
+//
+// Output:
+//      - out_vector    : Pointer to the last sample in a int16_t vector
+//                        of length |length|
+//
+
+//
+// WebRtcSpl_CopyFromEndW16(...)
+//
+// Copies the rightmost |samples| of |in_vector| (of length |in_vector_length|)
+// to the vector |out_vector|.
+//
+// Input:
+//      - in_vector         : Input vector
+//      - in_vector_length  : Number of samples in |in_vector|
+//      - samples           : Number of samples to extract (from right side)
+//                            from |in_vector|
+//
+// Output:
+//      - out_vector        : Vector with the requested samples
+//
+// Return value             : Number of copied samples in |out_vector|
+//
+
+//
+// WebRtcSpl_ZerosArrayW16(...)
+// WebRtcSpl_ZerosArrayW32(...)
+//
+// Inserts the value "zero" in all positions of a w16 and a w32 vector
+// respectively.
+//
+// Input:
+//      - vector_length : Number of samples in vector
+//
+// Output:
+//      - vector        : Vector containing all zeros
+//
+// Return value         : Number of samples in vector
+//
+
+//
+// WebRtcSpl_OnesArrayW16(...)
+// WebRtcSpl_OnesArrayW32(...)
+//
+// Inserts the value "one" in all positions of a w16 and a w32 vector
+// respectively.
+//
+// Input:
+//      - vector_length : Number of samples in vector
+//
+// Output:
+//      - vector        : Vector containing all ones
+//
+// Return value         : Number of samples in vector
+//
+
+//
+// WebRtcSpl_VectorBitShiftW16(...)
+// WebRtcSpl_VectorBitShiftW32(...)
+//
+// Bit shifts all the values in a vector up or downwards. Different calls for
+// int16_t and int32_t vectors respectively.
+//
+// Input:
+//      - vector_length : Length of vector
+//      - in_vector     : Pointer to the vector that should be bit shifted
+//      - right_shifts  : Number of right bit shifts (negative value gives left
+//                        shifts)
+//
+// Output:
+//      - out_vector    : Pointer to the result vector (can be the same as
+//                        |in_vector|)
+//
+
+//
+// WebRtcSpl_VectorBitShiftW32ToW16(...)
+//
+// Bit shifts all the values in a int32_t vector up or downwards and
+// stores the result as an int16_t vector. The function will saturate the
+// signal if needed, before storing in the output vector.
+//
+// Input:
+//      - vector_length : Length of vector
+//      - in_vector     : Pointer to the vector that should be bit shifted
+//      - right_shifts  : Number of right bit shifts (negative value gives left
+//                        shifts)
+//
+// Output:
+//      - out_vector    : Pointer to the result vector (can be the same as
+//                        |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleVector(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = (gain*in_vector[k])>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Scaling gain
+//      - vector_length : Elements in the |in_vector|
+//      - right_shifts  : Number of right bit shifts applied
+//
+// Output:
+//      - out_vector    : Output vector (can be the same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleVectorWithSat(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = SATURATE( (gain*in_vector[k])>>right_shifts )
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Scaling gain
+//      - vector_length : Elements in the |in_vector|
+//      - right_shifts  : Number of right bit shifts applied
+//
+// Output:
+//      - out_vector    : Output vector (can be the same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleAndAddVectors(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = (gain1*in_vector1[k])>>right_shifts1
+//                  + (gain2*in_vector2[k])>>right_shifts2
+//
+// Input:
+//      - in_vector1    : Input vector 1
+//      - gain1         : Gain to be used for vector 1
+//      - right_shifts1 : Right bit shift to be used for vector 1
+//      - in_vector2    : Input vector 2
+//      - gain2         : Gain to be used for vector 2
+//      - right_shifts2 : Right bit shift to be used for vector 2
+//      - vector_length : Elements in the input vectors
+//
+// Output:
+//      - out_vector    : Output vector
+//
+
+//
+// WebRtcSpl_ReverseOrderMultArrayElements(...)
+//
+// Performs the vector operation:
+//  out_vector[n] = (in_vector[n]*window[-n])>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - window        : Window vector (should be reversed). The pointer
+//                        should be set to the last value in the vector
+//      - right_shifts  : Number of right bit shift to be applied after the
+//                        multiplication
+//      - vector_length : Number of elements in |in_vector|
+//
+// Output:
+//      - out_vector    : Output vector (can be same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ElementwiseVectorMult(...)
+//
+// Performs the vector operation:
+//  out_vector[n] = (in_vector[n]*window[n])>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - window        : Window vector.
+//      - right_shifts  : Number of right bit shift to be applied after the
+//                        multiplication
+//      - vector_length : Number of elements in |in_vector|
+//
+// Output:
+//      - out_vector    : Output vector (can be same as |in_vector|)
+//
+
+//
+// WebRtcSpl_AddVectorsAndShift(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = (in_vector1[k] + in_vector2[k])>>right_shifts
+//
+// Input:
+//      - in_vector1    : Input vector 1
+//      - in_vector2    : Input vector 2
+//      - right_shifts  : Number of right bit shift to be applied after the
+//                        multiplication
+//      - vector_length : Number of elements in |in_vector1| and |in_vector2|
+//
+// Output:
+//      - out_vector    : Output vector (can be same as |in_vector1|)
+//
+
+//
+// WebRtcSpl_AddAffineVectorToVector(...)
+//
+// Adds an affine transformed vector to another vector |out_vector|, i.e,
+// performs
+//  out_vector[k] += (in_vector[k]*gain+add_constant)>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Gain value, used to multiply the in vector with
+//      - add_constant  : Constant value to add (usually 1<<(right_shifts-1),
+//                        but others can be used as well
+//      - right_shifts  : Number of right bit shifts (0-16)
+//      - vector_length : Number of samples in |in_vector| and |out_vector|
+//
+// Output:
+//      - out_vector    : Vector with the output
+//
+
+//
+// WebRtcSpl_AffineTransformVector(...)
+//
+// Affine transforms a vector, i.e, performs
+//  out_vector[k] = (in_vector[k]*gain+add_constant)>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Gain value, used to multiply the in vector with
+//      - add_constant  : Constant value to add (usually 1<<(right_shifts-1),
+//                        but others can be used as well
+//      - right_shifts  : Number of right bit shifts (0-16)
+//      - vector_length : Number of samples in |in_vector| and |out_vector|
+//
+// Output:
+//      - out_vector    : Vector with the output
+//
+
+//
+// WebRtcSpl_IncreaseSeed(...)
+//
+// Increases the seed (and returns the new value)
+//
+// Input:
+//      - seed      : Seed for random calculation
+//
+// Output:
+//      - seed      : Updated seed value
+//
+// Return value     : The new seed value
+//
+
+//
+// WebRtcSpl_RandU(...)
+//
+// Produces a uniformly distributed value in the int16_t range
+//
+// Input:
+//      - seed      : Seed for random calculation
+//
+// Output:
+//      - seed      : Updated seed value
+//
+// Return value     : Uniformly distributed value in the range
+//                    [Word16_MIN...Word16_MAX]
+//
+
+//
+// WebRtcSpl_RandN(...)
+//
+// Produces a normal distributed value in the int16_t range
+//
+// Input:
+//      - seed      : Seed for random calculation
+//
+// Output:
+//      - seed      : Updated seed value
+//
+// Return value     : N(0,1) value in the Q13 domain
+//
+
+//
+// WebRtcSpl_RandUArray(...)
+//
+// Produces a uniformly distributed vector with elements in the int16_t
+// range
+//
+// Input:
+//      - vector_length : Samples wanted in the vector
+//      - seed          : Seed for random calculation
+//
+// Output:
+//      - vector        : Vector with the uniform values
+//      - seed          : Updated seed value
+//
+// Return value         : Number of samples in vector, i.e., |vector_length|
+//
+
+//
+// WebRtcSpl_Sqrt(...)
+//
+// Returns the square root of the input value |value|. The precision of this
+// function is integer precision, i.e., sqrt(8) gives 2 as answer.
+// If |value| is a negative number then 0 is returned.
+//
+// Algorithm:
+//
+// A sixth order Taylor Series expansion is used here to compute the square
+// root of a number y^0.5 = (1+x)^0.5
+// where
+// x = y-1
+//   = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
+// 0.5 <= x < 1
+//
+// Input:
+//      - value     : Value to calculate sqrt of
+//
+// Return value     : Result of the sqrt calculation
+//
+
+//
+// WebRtcSpl_SqrtFloor(...)
+//
+// Returns the square root of the input value |value|. The precision of this
+// function is rounding down integer precision, i.e., sqrt(8) gives 2 as answer.
+// If |value| is a negative number then 0 is returned.
+//
+// Algorithm:
+//
+// An iterative 4 cylce/bit routine
+//
+// Input:
+//      - value     : Value to calculate sqrt of
+//
+// Return value     : Result of the sqrt calculation
+//
+
+//
+// WebRtcSpl_DivU32U16(...)
+//
+// Divides a uint32_t |num| by a uint16_t |den|.
+//
+// If |den|==0, (uint32_t)0xFFFFFFFF is returned.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division (as a uint32_t), i.e., the
+//                    integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivW32W16(...)
+//
+// Divides a int32_t |num| by a int16_t |den|.
+//
+// If |den|==0, (int32_t)0x7FFFFFFF is returned.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division (as a int32_t), i.e., the
+//                    integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivW32W16ResW16(...)
+//
+// Divides a int32_t |num| by a int16_t |den|, assuming that the
+// result is less than 32768, otherwise an unpredictable result will occur.
+//
+// If |den|==0, (int16_t)0x7FFF is returned.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division (as a int16_t), i.e., the
+//                    integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivResultInQ31(...)
+//
+// Divides a int32_t |num| by a int16_t |den|, assuming that the
+// absolute value of the denominator is larger than the numerator, otherwise
+// an unpredictable result will occur.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division in Q31.
+//
+
+//
+// WebRtcSpl_DivW32HiLow(...)
+//
+// Divides a int32_t |num| by a denominator in hi, low format. The
+// absolute value of the denominator has to be larger (or equal to) the
+// numerator.
+//
+// Input:
+//      - num       : Numerator
+//      - den_hi    : High part of denominator
+//      - den_low   : Low part of denominator
+//
+// Return value     : Divided value in Q31
+//
+
+//
+// WebRtcSpl_Energy(...)
+//
+// Calculates the energy of a vector
+//
+// Input:
+//      - vector        : Vector which the energy should be calculated on
+//      - vector_length : Number of samples in vector
+//
+// Output:
+//      - scale_factor  : Number of left bit shifts needed to get the physical
+//                        energy value, i.e, to get the Q0 value
+//
+// Return value         : Energy value in Q(-|scale_factor|)
+//
+
+//
+// WebRtcSpl_FilterAR(...)
+//
+// Performs a 32-bit AR filtering on a vector in Q12
+//
+// Input:
+//  - ar_coef                   : AR-coefficient vector (values in Q12),
+//                                ar_coef[0] must be 4096.
+//  - ar_coef_length            : Number of coefficients in |ar_coef|.
+//  - in_vector                 : Vector to be filtered.
+//  - in_vector_length          : Number of samples in |in_vector|.
+//  - filter_state              : Current state (higher part) of the filter.
+//  - filter_state_length       : Length (in samples) of |filter_state|.
+//  - filter_state_low          : Current state (lower part) of the filter.
+//  - filter_state_low_length   : Length (in samples) of |filter_state_low|.
+//  - out_vector_low_length     : Maximum length (in samples) of
+//                                |out_vector_low|.
+//
+// Output:
+//  - filter_state              : Updated state (upper part) vector.
+//  - filter_state_low          : Updated state (lower part) vector.
+//  - out_vector                : Vector containing the upper part of the
+//                                filtered values.
+//  - out_vector_low            : Vector containing the lower part of the
+//                                filtered values.
+//
+// Return value                 : Number of samples in the |out_vector|.
+//
+
+//
+// WebRtcSpl_FilterMAFastQ12(...)
+//
+// Performs a MA filtering on a vector in Q12
+//
+// Input:
+//      - in_vector         : Input samples (state in positions
+//                            in_vector[-order] .. in_vector[-1])
+//      - ma_coef           : Filter coefficients (in Q12)
+//      - ma_coef_length    : Number of B coefficients (order+1)
+//      - vector_length     : Number of samples to be filtered
+//
+// Output:
+//      - out_vector        : Filtered samples
+//
+
+//
+// WebRtcSpl_ComplexIFFT(...)
+//
+// Complex Inverse FFT
+//
+// Computes an inverse complex 2^|stages|-point FFT on the input vector, which
+// is in bit-reversed order. The original content of the vector is destroyed in
+// the process, since the input is overwritten by the output, normal-ordered,
+// FFT vector. With X as the input complex vector, y as the output complex
+// vector and with M = 2^|stages|, the following is computed:
+//
+//        M-1
+// y(k) = sum[X(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
+//        i=0
+//
+// The implementations are optimized for speed, not for code size. It uses the
+// decimation-in-time algorithm with radix-2 butterfly technique.
+//
+// Input:
+//      - vector    : In pointer to complex vector containing 2^|stages|
+//                    real elements interleaved with 2^|stages| imaginary
+//                    elements.
+//                    [ReImReImReIm....]
+//                    The elements are in Q(-scale) domain, see more on Return
+//                    Value below.
+//
+//      - stages    : Number of FFT stages. Must be at least 3 and at most 10,
+//                    since the table WebRtcSpl_kSinTable1024[] is 1024
+//                    elements long.
+//
+//      - mode      : This parameter gives the user to choose how the FFT
+//                    should work.
+//                    mode==0: Low-complexity and Low-accuracy mode
+//                    mode==1: High-complexity and High-accuracy mode
+//
+// Output:
+//      - vector    : Out pointer to the FFT vector (the same as input).
+//
+// Return Value     : The scale value that tells the number of left bit shifts
+//                    that the elements in the |vector| should be shifted with
+//                    in order to get Q0 values, i.e. the physically correct
+//                    values. The scale parameter is always 0 or positive,
+//                    except if N>1024 (|stages|>10), which returns a scale
+//                    value of -1, indicating error.
+//
+
+//
+// WebRtcSpl_ComplexFFT(...)
+//
+// Complex FFT
+//
+// Computes a complex 2^|stages|-point FFT on the input vector, which is in
+// bit-reversed order. The original content of the vector is destroyed in
+// the process, since the input is overwritten by the output, normal-ordered,
+// FFT vector. With x as the input complex vector, Y as the output complex
+// vector and with M = 2^|stages|, the following is computed:
+//
+//              M-1
+// Y(k) = 1/M * sum[x(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
+//              i=0
+//
+// The implementations are optimized for speed, not for code size. It uses the
+// decimation-in-time algorithm with radix-2 butterfly technique.
+//
+// This routine prevents overflow by scaling by 2 before each FFT stage. This is
+// a fixed scaling, for proper normalization - there will be log2(n) passes, so
+// this results in an overall factor of 1/n, distributed to maximize arithmetic
+// accuracy.
+//
+// Input:
+//      - vector    : In pointer to complex vector containing 2^|stages| real
+//                    elements interleaved with 2^|stages| imaginary elements.
+//                    [ReImReImReIm....]
+//                    The output is in the Q0 domain.
+//
+//      - stages    : Number of FFT stages. Must be at least 3 and at most 10,
+//                    since the table WebRtcSpl_kSinTable1024[] is 1024
+//                    elements long.
+//
+//      - mode      : This parameter gives the user to choose how the FFT
+//                    should work.
+//                    mode==0: Low-complexity and Low-accuracy mode
+//                    mode==1: High-complexity and High-accuracy mode
+//
+// Output:
+//      - vector    : The output FFT vector is in the Q0 domain.
+//
+// Return value     : The scale parameter is always 0, except if N>1024,
+//                    which returns a scale value of -1, indicating error.
+//
+
+//
+// WebRtcSpl_AnalysisQMF(...)
+//
+// Splits a 0-2*F Hz signal into two sub bands: 0-F Hz and F-2*F Hz. The
+// current version has F = 8000, therefore, a super-wideband audio signal is
+// split to lower-band 0-8 kHz and upper-band 8-16 kHz.
+//
+// Input:
+//      - in_data       : Wide band speech signal, 320 samples (10 ms)
+//
+// Input & Output:
+//      - filter_state1 : Filter state for first All-pass filter
+//      - filter_state2 : Filter state for second All-pass filter
+//
+// Output:
+//      - low_band      : Lower-band signal 0-8 kHz band, 160 samples (10 ms)
+//      - high_band     : Upper-band signal 8-16 kHz band (flipped in frequency
+//                        domain), 160 samples (10 ms)
+//
+
+//
+// WebRtcSpl_SynthesisQMF(...)
+//
+// Combines the two sub bands (0-F and F-2*F Hz) into a signal of 0-2*F
+// Hz, (current version has F = 8000 Hz). So the filter combines lower-band
+// (0-8 kHz) and upper-band (8-16 kHz) channels to obtain super-wideband 0-16
+// kHz audio.
+//
+// Input:
+//      - low_band      : The signal with the 0-8 kHz band, 160 samples (10 ms)
+//      - high_band     : The signal with the 8-16 kHz band, 160 samples (10 ms)
+//
+// Input & Output:
+//      - filter_state1 : Filter state for first All-pass filter
+//      - filter_state2 : Filter state for second All-pass filter
+//
+// Output:
+//      - out_data      : Super-wideband speech signal, 0-16 kHz
+//
+
+// int16_t WebRtcSpl_SatW32ToW16(...)
+//
+// This function saturates a 32-bit word into a 16-bit word.
+//
+// Input:
+//      - value32   : The value of a 32-bit word.
+//
+// Output:
+//      - out16     : the saturated 16-bit word.
+//
+
+// int32_t WebRtc_MulAccumW16(...)
+//
+// This function multiply a 16-bit word by a 16-bit word, and accumulate this
+// value to a 32-bit integer.
+//
+// Input:
+//      - a    : The value of the first 16-bit word.
+//      - b    : The value of the second 16-bit word.
+//      - c    : The value of an 32-bit integer.
+//
+// Return Value: The value of a * b + c.
+//
+
+// int16_t WebRtcSpl_get_version(...)
+//
+// This function gives the version string of the Signal Processing Library.
+//
+// Input:
+//      - length_in_bytes : The size of Allocated space (in Bytes) where
+//                          the version number is written to (in string format).
+//
+// Output:
+//      - version         : Pointer to a buffer where the version number is
+//                          written to.
+//
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl.h
new file mode 100644
index 0000000..6366f10
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl.h
@@ -0,0 +1,177 @@
+/*
+ *  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_V7
+#include "include/spl_inl_armv7.h"
+#else
+
+#if defined(MIPS32_LE)
+#include "include/spl_inl_mips.h"
+#endif
+
+#if !defined(MIPS_DSP_R1_LE)
+static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
+  int16_t out16 = (int16_t) value32;
+
+  if (value32 > 32767)
+    out16 = 32767;
+  else if (value32 < -32768)
+    out16 = -32768;
+
+  return out16;
+}
+
+static __inline int16_t WebRtcSpl_AddSatW16(int16_t a, int16_t b) {
+  return WebRtcSpl_SatW32ToW16((int32_t) a + (int32_t) b);
+}
+
+static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
+  return WebRtcSpl_SatW32ToW16((int32_t) var1 - (int32_t) var2);
+}
+#endif  // #if !defined(MIPS_DSP_R1_LE)
+
+#if !defined(MIPS32_LE)
+static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t 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(int32_t a) {
+  int zeros;
+
+  if (a == 0) {
+    return 0;
+  }
+  else if (a < 0) {
+    a = ~a;
+  }
+
+  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(uint32_t 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(int16_t a) {
+  int zeros;
+
+  if (a == 0) {
+    return 0;
+  }
+  else if (a < 0) {
+    a = ~a;
+  }
+
+  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  // #if !defined(MIPS32_LE)
+
+#endif  // WEBRTC_ARCH_ARM_V7
+
+// The following functions have no optimized versions.
+// TODO(kma): Consider saturating add/sub instructions in X86 platform.
+#if !defined(MIPS_DSP_R1_LE)
+static __inline int32_t WebRtcSpl_AddSatW32(int32_t l_var1, int32_t l_var2) {
+  int32_t l_sum;
+
+  // Perform long addition
+  l_sum = l_var1 + l_var2;
+
+  if (l_var1 < 0) {  // Check for underflow.
+    if ((l_var2 < 0) && (l_sum >= 0)) {
+        l_sum = (int32_t)0x80000000;
+    }
+  } else {  // Check for overflow.
+    if ((l_var2 > 0) && (l_sum < 0)) {
+        l_sum = (int32_t)0x7FFFFFFF;
+    }
+  }
+
+  return l_sum;
+}
+
+static __inline int32_t WebRtcSpl_SubSatW32(int32_t l_var1, int32_t l_var2) {
+  int32_t l_diff;
+
+  // Perform subtraction.
+  l_diff = l_var1 - l_var2;
+
+  if (l_var1 < 0) {  // Check for underflow.
+    if ((l_var2 > 0) && (l_diff > 0)) {
+      l_diff = (int32_t)0x80000000;
+    }
+  } else {  // Check for overflow.
+    if ((l_var2 < 0) && (l_diff < 0)) {
+      l_diff = (int32_t)0x7FFFFFFF;
+    }
+  }
+
+  return l_diff;
+}
+#endif  // #if !defined(MIPS_DSP_R1_LE)
+
+#endif  // WEBRTC_SPL_SPL_INL_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h
new file mode 100644
index 0000000..fdbcb43
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h
@@ -0,0 +1,165 @@
+/*
+ *  Copyright (c) 2012 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_
+
+/* TODO(kma): Replace some assembly code with GCC intrinsics
+ * (e.g. __builtin_clz).
+ */
+
+/* This function produces result that is not bit exact with that by the generic
+ * C version in some cases, although the former is at least as accurate as the
+ * later.
+ */
+static __inline int32_t WEBRTC_SPL_MUL_16_32_RSFT16(int16_t a, int32_t b) {
+  int32_t tmp = 0;
+  __asm __volatile ("smulwb %0, %1, %2":"=r"(tmp):"r"(b), "r"(a));
+  return tmp;
+}
+
+/* This function produces result that is not bit exact with that by the generic
+ * C version in some cases, although the former is at least as accurate as the
+ * later.
+ */
+static __inline int32_t WEBRTC_SPL_MUL_32_32_RSFT32(int16_t a,
+                                                    int16_t b,
+                                                    int32_t c) {
+  int32_t tmp = 0;
+  __asm __volatile (
+    "pkhbt %[tmp], %[b], %[a], lsl #16\n\t"
+    "smmulr %[tmp], %[tmp], %[c]\n\t"
+    :[tmp]"+r"(tmp)
+    :[a]"r"(a),
+     [b]"r"(b),
+     [c]"r"(c)
+  );
+  return tmp;
+}
+
+static __inline int32_t WEBRTC_SPL_MUL_32_32_RSFT32BI(int32_t a, int32_t b) {
+  int32_t tmp = 0;
+  __asm volatile ("smmulr %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
+  return tmp;
+}
+
+static __inline int32_t WEBRTC_SPL_MUL_16_16(int16_t a, int16_t b) {
+  int32_t tmp = 0;
+  __asm __volatile ("smulbb %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
+  return tmp;
+}
+
+// TODO(kma): add unit test.
+static __inline int32_t WebRtc_MulAccumW16(int16_t a, int16_t b, int32_t c) {
+  int32_t tmp = 0;
+  __asm __volatile ("smlabb %0, %1, %2, %3":"=r"(tmp):"r"(a), "r"(b), "r"(c));
+  return tmp;
+}
+
+static __inline int16_t WebRtcSpl_AddSatW16(int16_t a, int16_t b) {
+  int32_t s_sum = 0;
+
+  __asm __volatile ("qadd16 %0, %1, %2":"=r"(s_sum):"r"(a), "r"(b));
+
+  return (int16_t) s_sum;
+}
+
+/* TODO(kma): find the cause of unittest errors by the next two functions:
+ * http://code.google.com/p/webrtc/issues/detail?id=740.
+ */
+#if 0
+static __inline int32_t WebRtcSpl_AddSatW32(int32_t l_var1, int32_t l_var2) {
+  int32_t l_sum = 0;
+
+  __asm __volatile ("qadd %0, %1, %2":"=r"(l_sum):"r"(l_var1), "r"(l_var2));
+
+  return l_sum;
+}
+
+static __inline int32_t WebRtcSpl_SubSatW32(int32_t l_var1, int32_t l_var2) {
+  int32_t l_sub = 0;
+
+  __asm __volatile ("qsub %0, %1, %2":"=r"(l_sub):"r"(l_var1), "r"(l_var2));
+
+  return l_sub;
+}
+#endif
+
+static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
+  int32_t s_sub = 0;
+
+  __asm __volatile ("qsub16 %0, %1, %2":"=r"(s_sub):"r"(var1), "r"(var2));
+
+  return (int16_t)s_sub;
+}
+
+static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
+  int32_t tmp = 0;
+
+  __asm __volatile ("clz %0, %1":"=r"(tmp):"r"(n));
+
+  return (int16_t)(32 - tmp);
+}
+
+static __inline int WebRtcSpl_NormW32(int32_t a) {
+  int32_t tmp = 0;
+
+  if (a == 0) {
+    return 0;
+  }
+  else if (a < 0) {
+    a ^= 0xFFFFFFFF;
+  }
+
+  __asm __volatile ("clz %0, %1":"=r"(tmp):"r"(a));
+
+  return tmp - 1;
+}
+
+static __inline int WebRtcSpl_NormU32(uint32_t a) {
+  int tmp = 0;
+
+  if (a == 0) return 0;
+
+  __asm __volatile ("clz %0, %1":"=r"(tmp):"r"(a));
+
+  return tmp;
+}
+
+static __inline int WebRtcSpl_NormW16(int16_t a) {
+  int32_t tmp = 0;
+
+  if (a == 0) {
+    return 0;
+  }
+  else if (a < 0) {
+    a ^= 0xFFFFFFFF;
+  }
+
+  __asm __volatile ("clz %0, %1":"=r"(tmp):"r"(a));
+
+  return tmp - 17;
+}
+
+// TODO(kma): add unit test.
+static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
+  int32_t out = 0;
+
+  __asm __volatile ("ssat %0, #16, %1" : "=r"(out) : "r"(value32));
+
+  return (int16_t)out;
+}
+
+#endif  // WEBRTC_SPL_SPL_INL_ARMV7_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl_mips.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl_mips.h
new file mode 100644
index 0000000..e173777
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/include/spl_inl_mips.h
@@ -0,0 +1,281 @@
+/*
+ *  Copyright (c) 2013 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_MIPS_H_
+#define WEBRTC_SPL_SPL_INL_MIPS_H_
+
+static __inline int32_t WEBRTC_SPL_MUL_16_16(int32_t a,
+                                             int32_t b) {
+  int32_t value32 = 0;
+  int32_t a1 = 0, b1 = 0;
+
+  __asm __volatile(
+#if defined(MIPS32_R2_LE)
+    "seh    %[a1],          %[a]                \n\t"
+    "seh    %[b1],          %[b]                \n\t"
+#else
+    "sll    %[a1],          %[a],         16    \n\t"
+    "sll    %[b1],          %[b],         16    \n\t"
+    "sra    %[a1],          %[a1],        16    \n\t"
+    "sra    %[b1],          %[b1],        16    \n\t"
+#endif
+    "mul    %[value32],     %[a1],  %[b1]       \n\t"
+    : [value32] "=r" (value32), [a1] "=&r" (a1), [b1] "=&r" (b1)
+    : [a] "r" (a), [b] "r" (b)
+    : "hi", "lo"
+  );
+  return value32;
+}
+
+static __inline int32_t WEBRTC_SPL_MUL_16_32_RSFT16(int16_t a,
+                                                    int32_t b) {
+  int32_t value32 = 0, b1 = 0, b2 = 0;
+  int32_t a1 = 0;
+
+  __asm __volatile(
+#if defined(MIPS32_R2_LE)
+    "seh    %[a1],          %[a]                        \n\t"
+#else
+    "sll    %[a1],          %[a],           16          \n\t"
+    "sra    %[a1],          %[a1],          16          \n\t"
+#endif
+    "andi   %[b2],          %[b],           0xFFFF      \n\t"
+    "sra    %[b1],          %[b],           16          \n\t"
+    "sra    %[b2],          %[b2],          1           \n\t"
+    "mul    %[value32],     %[a1],          %[b1]       \n\t"
+    "mul    %[b2],          %[a1],          %[b2]       \n\t"
+    "addiu  %[b2],          %[b2],          0x4000      \n\t"
+    "sra    %[b2],          %[b2],          15          \n\t"
+    "addu   %[value32],     %[value32],     %[b2]       \n\t"
+    : [value32] "=&r" (value32), [b1] "=&r" (b1), [b2] "=&r" (b2),
+      [a1] "=&r" (a1)
+    : [a] "r" (a), [b] "r" (b)
+    : "hi", "lo"
+  );
+  return value32;
+}
+
+static __inline int32_t WEBRTC_SPL_MUL_32_32_RSFT32BI(int32_t a,
+                                                      int32_t b) {
+  int32_t tmp = 0;
+
+  if ((32767 < a) || (a < 0))
+    tmp = WEBRTC_SPL_MUL_16_32_RSFT16(((int16_t)(a >> 16)), b);
+  tmp += WEBRTC_SPL_MUL_16_32_RSFT16(((int16_t)((a & 0x0000FFFF) >> 1)),
+                                     b) >> 15;
+
+  return tmp;
+}
+
+static __inline int32_t WEBRTC_SPL_MUL_32_32_RSFT32(int16_t a,
+                                                    int16_t b,
+                                                    int32_t c) {
+  int32_t tmp1 = 0, tmp2 = 0, tmp3 = 0, tmp4 = 0;
+
+  __asm __volatile(
+    "sra                %[tmp1],   %[c],      16      \n\t"
+    "andi               %[tmp2],   %[c],      0xFFFF  \n\t"
+#if defined(MIPS32_R2_LE)
+    "seh                %[a],      %[a]               \n\t"
+    "seh                %[b],      %[b]               \n\t"
+#else
+    "sll                %[a],      %[a],      16      \n\t"
+    "sra                %[a],      %[a],      16      \n\t"
+    "sll                %[b],      %[b],      16      \n\t"
+    "sra                %[b],      %[b],      16      \n\t"
+#endif
+    "sra                %[tmp2],   %[tmp2],   1       \n\t"
+    "mul                %[tmp3],   %[a],      %[tmp2] \n\t"
+    "mul                %[tmp4],   %[b],      %[tmp2] \n\t"
+    "mul                %[tmp2],   %[a],      %[tmp1] \n\t"
+    "mul                %[tmp1],   %[b],      %[tmp1] \n\t"
+#if defined(MIPS_DSP_R1_LE)
+    "shra_r.w           %[tmp3],   %[tmp3],   15      \n\t"
+    "shra_r.w           %[tmp4],   %[tmp4],   15      \n\t"
+#else
+    "addiu              %[tmp3],   %[tmp3],   0x4000  \n\t"
+    "sra                %[tmp3],   %[tmp3],   15      \n\t"
+    "addiu              %[tmp4],   %[tmp4],   0x4000  \n\t"
+    "sra                %[tmp4],   %[tmp4],   15      \n\t"
+#endif
+    "addu               %[tmp3],   %[tmp3],   %[tmp2] \n\t"
+    "addu               %[tmp4],   %[tmp4],   %[tmp1] \n\t"
+    "sra                %[tmp4],   %[tmp4],   16      \n\t"
+    "addu               %[tmp1],   %[tmp3],   %[tmp4] \n\t"
+    : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2),
+      [tmp3] "=&r" (tmp3), [tmp4] "=&r" (tmp4),
+      [a] "+r" (a), [b] "+r" (b)
+    : [c] "r" (c)
+    : "hi", "lo"
+  );
+  return tmp1;
+}
+
+#if defined(MIPS_DSP_R1_LE)
+static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
+  __asm __volatile(
+    "shll_s.w   %[value32], %[value32], 16      \n\t"
+    "sra        %[value32], %[value32], 16      \n\t"
+    : [value32] "+r" (value32)
+    :
+  );
+  int16_t out16 = (int16_t)value32;
+  return out16;
+}
+
+static __inline int16_t WebRtcSpl_AddSatW16(int16_t a, int16_t b) {
+  int32_t value32 = 0;
+
+  __asm __volatile(
+    "addq_s.ph      %[value32],     %[a],   %[b]    \n\t"
+    : [value32] "=r" (value32)
+    : [a] "r" (a), [b] "r" (b)
+  );
+  return (int16_t)value32;
+}
+
+static __inline int32_t WebRtcSpl_AddSatW32(int32_t l_var1, int32_t l_var2) {
+  int32_t l_sum;
+
+  __asm __volatile(
+    "addq_s.w   %[l_sum],       %[l_var1],      %[l_var2]    \n\t"
+    : [l_sum] "=r" (l_sum)
+    : [l_var1] "r" (l_var1), [l_var2] "r" (l_var2)
+  );
+
+  return l_sum;
+}
+
+static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
+  int32_t value32;
+
+  __asm __volatile(
+    "subq_s.ph  %[value32], %[var1],    %[var2]     \n\t"
+    : [value32] "=r" (value32)
+    : [var1] "r" (var1), [var2] "r" (var2)
+  );
+
+  return (int16_t)value32;
+}
+
+static __inline int32_t WebRtcSpl_SubSatW32(int32_t l_var1, int32_t l_var2) {
+  int32_t l_diff;
+
+  __asm __volatile(
+    "subq_s.w   %[l_diff],      %[l_var1],      %[l_var2]    \n\t"
+    : [l_diff] "=r" (l_diff)
+    : [l_var1] "r" (l_var1), [l_var2] "r" (l_var2)
+  );
+
+  return l_diff;
+}
+#endif
+
+static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
+  int bits = 0;
+  int i32 = 32;
+
+  __asm __volatile(
+    "clz    %[bits],    %[n]                    \n\t"
+    "subu   %[bits],    %[i32],     %[bits]     \n\t"
+    : [bits] "=&r" (bits)
+    : [n] "r" (n), [i32] "r" (i32)
+  );
+
+  return bits;
+}
+
+static __inline int WebRtcSpl_NormW32(int32_t a) {
+  int zeros = 0;
+
+  __asm __volatile(
+    ".set       push                                \n\t"
+    ".set       noreorder                           \n\t"
+    "bnez       %[a],       1f                      \n\t"
+    " sra       %[zeros],   %[a],       31          \n\t"
+    "b          2f                                  \n\t"
+    " move      %[zeros],   $zero                   \n\t"
+   "1:                                              \n\t"
+    "xor        %[zeros],   %[a],       %[zeros]    \n\t"
+    "clz        %[zeros],   %[zeros]                \n\t"
+    "addiu      %[zeros],   %[zeros],   -1          \n\t"
+   "2:                                              \n\t"
+    ".set       pop                                 \n\t"
+    : [zeros]"=&r"(zeros)
+    : [a] "r" (a)
+  );
+
+  return zeros;
+}
+
+static __inline int WebRtcSpl_NormU32(uint32_t a) {
+  int zeros = 0;
+
+  __asm __volatile(
+    "clz    %[zeros],   %[a]    \n\t"
+    : [zeros] "=r" (zeros)
+    : [a] "r" (a)
+  );
+
+  return (zeros & 0x1f);
+}
+
+static __inline int WebRtcSpl_NormW16(int16_t a) {
+  int zeros = 0;
+  int a0 = a << 16;
+
+  __asm __volatile(
+    ".set       push                                \n\t"
+    ".set       noreorder                           \n\t"
+    "bnez       %[a0],      1f                      \n\t"
+    " sra       %[zeros],   %[a0],      31          \n\t"
+    "b          2f                                  \n\t"
+    " move      %[zeros],   $zero                   \n\t"
+   "1:                                              \n\t"
+    "xor        %[zeros],   %[a0],      %[zeros]    \n\t"
+    "clz        %[zeros],   %[zeros]                \n\t"
+    "addiu      %[zeros],   %[zeros],   -1          \n\t"
+   "2:                                              \n\t"
+    ".set       pop                                 \n\t"
+    : [zeros]"=&r"(zeros)
+    : [a0] "r" (a0)
+  );
+
+  return zeros;
+}
+
+static __inline int32_t WebRtc_MulAccumW16(int16_t a,
+                                           int16_t b,
+                                           int32_t c) {
+  int32_t res = 0, c1 = 0;
+  __asm __volatile(
+#if defined(MIPS32_R2_LE)
+    "seh    %[a],       %[a]            \n\t"
+    "seh    %[b],       %[b]            \n\t"
+#else
+    "sll    %[a],       %[a],   16      \n\t"
+    "sll    %[b],       %[b],   16      \n\t"
+    "sra    %[a],       %[a],   16      \n\t"
+    "sra    %[b],       %[b],   16      \n\t"
+#endif
+    "mul    %[res],     %[a],   %[b]    \n\t"
+    "addu   %[c1],      %[c],   %[res]  \n\t"
+    : [c1] "=r" (c1), [res] "=&r" (res)
+    : [a] "r" (a), [b] "r" (b), [c] "r" (c)
+    : "hi", "lo"
+  );
+  return (c1);
+}
+
+#endif  // WEBRTC_SPL_SPL_INL_MIPS_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/min_max_operations.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/min_max_operations.c
new file mode 100644
index 0000000..581272a
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/min_max_operations.c
@@ -0,0 +1,243 @@
+/*
+ *  Copyright (c) 2012 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_MaxAbsValueW16C()
+ * WebRtcSpl_MaxAbsValueW32C()
+ * WebRtcSpl_MaxValueW16C()
+ * WebRtcSpl_MaxValueW32C()
+ * WebRtcSpl_MinValueW16C()
+ * WebRtcSpl_MinValueW32C()
+ * WebRtcSpl_MaxAbsIndexW16()
+ * WebRtcSpl_MaxIndexW16()
+ * WebRtcSpl_MaxIndexW32()
+ * WebRtcSpl_MinIndexW16()
+ * WebRtcSpl_MinIndexW32()
+ *
+ */
+
+#include "include/signal_processing_library.h"
+
+#include <stdlib.h>
+
+// TODO(bjorn/kma): Consolidate function pairs (e.g. combine
+//   WebRtcSpl_MaxAbsValueW16C and WebRtcSpl_MaxAbsIndexW16 into a single one.)
+// TODO(kma): Move the next six functions into min_max_operations_c.c.
+
+// Maximum absolute value of word16 vector. C version for generic platforms.
+int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, int length) {
+  int i = 0, absolute = 0, maximum = 0;
+
+  if (vector == NULL || length <= 0) {
+    return -1;
+  }
+
+  for (i = 0; i < length; i++) {
+    absolute = abs((int)vector[i]);
+
+    if (absolute > maximum) {
+      maximum = absolute;
+    }
+  }
+
+  // Guard the case for abs(-32768).
+  if (maximum > WEBRTC_SPL_WORD16_MAX) {
+    maximum = WEBRTC_SPL_WORD16_MAX;
+  }
+
+  return (int16_t)maximum;
+}
+
+// Maximum absolute value of word32 vector. C version for generic platforms.
+int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, int length) {
+  // Use uint32_t for the local variables, to accommodate the return value
+  // of abs(0x80000000), which is 0x80000000.
+
+  uint32_t absolute = 0, maximum = 0;
+  int i = 0;
+
+  if (vector == NULL || length <= 0) {
+    return -1;
+  }
+
+  for (i = 0; i < length; i++) {
+    absolute = abs((int)vector[i]);
+    if (absolute > maximum) {
+      maximum = absolute;
+    }
+  }
+
+  maximum = WEBRTC_SPL_MIN(maximum, WEBRTC_SPL_WORD32_MAX);
+
+  return (int32_t)maximum;
+}
+
+// Maximum value of word16 vector. C version for generic platforms.
+int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, int length) {
+  int16_t maximum = WEBRTC_SPL_WORD16_MIN;
+  int i = 0;
+
+  if (vector == NULL || length <= 0) {
+    return maximum;
+  }
+
+  for (i = 0; i < length; i++) {
+    if (vector[i] > maximum)
+      maximum = vector[i];
+  }
+  return maximum;
+}
+
+// Maximum value of word32 vector. C version for generic platforms.
+int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, int length) {
+  int32_t maximum = WEBRTC_SPL_WORD32_MIN;
+  int i = 0;
+
+  if (vector == NULL || length <= 0) {
+    return maximum;
+  }
+
+  for (i = 0; i < length; i++) {
+    if (vector[i] > maximum)
+      maximum = vector[i];
+  }
+  return maximum;
+}
+
+// Minimum value of word16 vector. C version for generic platforms.
+int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, int length) {
+  int16_t minimum = WEBRTC_SPL_WORD16_MAX;
+  int i = 0;
+
+  if (vector == NULL || length <= 0) {
+    return minimum;
+  }
+
+  for (i = 0; i < length; i++) {
+    if (vector[i] < minimum)
+      minimum = vector[i];
+  }
+  return minimum;
+}
+
+// Minimum value of word32 vector. C version for generic platforms.
+int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, int length) {
+  int32_t minimum = WEBRTC_SPL_WORD32_MAX;
+  int i = 0;
+
+  if (vector == NULL || length <= 0) {
+    return minimum;
+  }
+
+  for (i = 0; i < length; i++) {
+    if (vector[i] < minimum)
+      minimum = vector[i];
+  }
+  return minimum;
+}
+
+// Index of maximum absolute value in a word16 vector.
+int WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, int length) {
+  // Use type int for local variables, to accomodate the value of abs(-32768).
+
+  int i = 0, absolute = 0, maximum = 0, index = 0;
+
+  if (vector == NULL || length <= 0) {
+    return -1;
+  }
+
+  for (i = 0; i < length; i++) {
+    absolute = abs((int)vector[i]);
+
+    if (absolute > maximum) {
+      maximum = absolute;
+      index = i;
+    }
+  }
+
+  return index;
+}
+
+// Index of maximum value in a word16 vector.
+int WebRtcSpl_MaxIndexW16(const int16_t* vector, int length) {
+  int i = 0, index = 0;
+  int16_t maximum = WEBRTC_SPL_WORD16_MIN;
+
+  if (vector == NULL || length <= 0) {
+    return -1;
+  }
+
+  for (i = 0; i < length; i++) {
+    if (vector[i] > maximum) {
+      maximum = vector[i];
+      index = i;
+    }
+  }
+
+  return index;
+}
+
+// Index of maximum value in a word32 vector.
+int WebRtcSpl_MaxIndexW32(const int32_t* vector, int length) {
+  int i = 0, index = 0;
+  int32_t maximum = WEBRTC_SPL_WORD32_MIN;
+
+  if (vector == NULL || length <= 0) {
+    return -1;
+  }
+
+  for (i = 0; i < length; i++) {
+    if (vector[i] > maximum) {
+      maximum = vector[i];
+      index = i;
+    }
+  }
+
+  return index;
+}
+
+// Index of minimum value in a word16 vector.
+int WebRtcSpl_MinIndexW16(const int16_t* vector, int length) {
+  int i = 0, index = 0;
+  int16_t minimum = WEBRTC_SPL_WORD16_MAX;
+
+  if (vector == NULL || length <= 0) {
+    return -1;
+  }
+
+  for (i = 0; i < length; i++) {
+    if (vector[i] < minimum) {
+      minimum = vector[i];
+      index = i;
+    }
+  }
+
+  return index;
+}
+
+// Index of minimum value in a word32 vector.
+int WebRtcSpl_MinIndexW32(const int32_t* vector, int length) {
+  int i = 0, index = 0;
+  int32_t minimum = WEBRTC_SPL_WORD32_MAX;
+
+  if (vector == NULL || length <= 0) {
+    return -1;
+  }
+
+  for (i = 0; i < length; i++) {
+    if (vector[i] < minimum) {
+      minimum = vector[i];
+      index = i;
+    }
+  }
+
+  return index;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/real_fft.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/real_fft.c
new file mode 100644
index 0000000..006228c
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/real_fft.c
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2012 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 "include/real_fft.h"
+
+#include <stdlib.h>
+
+#include "include/signal_processing_library.h"
+
+struct RealFFT {
+  int order;
+};
+
+struct RealFFT* WebRtcSpl_CreateRealFFTC(int order) {
+  struct RealFFT* self = NULL;
+
+  if (order > kMaxFFTOrder || order < 0) {
+    return NULL;
+  }
+
+  self = malloc(sizeof(struct RealFFT));
+  if (self == NULL) {
+    return NULL;
+  }
+  self->order = order;
+
+  return self;
+}
+
+void WebRtcSpl_FreeRealFFTC(struct RealFFT* self) {
+  if (self != NULL) {
+    free(self);
+  }
+}
+
+// The C version FFT functions (i.e. WebRtcSpl_RealForwardFFTC and
+// WebRtcSpl_RealInverseFFTC) are real-valued FFT wrappers for complex-valued
+// FFT implementation in SPL.
+
+int WebRtcSpl_RealForwardFFTC(struct RealFFT* self,
+                              const int16_t* real_data_in,
+                              int16_t* complex_data_out) {
+  int i = 0;
+  int j = 0;
+  int result = 0;
+  int n = 1 << self->order;
+  // The complex-value FFT implementation needs a buffer to hold 2^order
+  // 16-bit COMPLEX numbers, for both time and frequency data.
+  int16_t complex_buffer[2 << kMaxFFTOrder];
+
+  // Insert zeros to the imaginary parts for complex forward FFT input.
+  for (i = 0, j = 0; i < n; i += 1, j += 2) {
+    complex_buffer[j] = real_data_in[i];
+    complex_buffer[j + 1] = 0;
+  };
+
+  WebRtcSpl_ComplexBitReverse(complex_buffer, self->order);
+  result = WebRtcSpl_ComplexFFT(complex_buffer, self->order, 1);
+
+  // For real FFT output, use only the first N + 2 elements from
+  // complex forward FFT.
+  memcpy(complex_data_out, complex_buffer, sizeof(int16_t) * (n + 2));
+
+  return result;
+}
+
+int WebRtcSpl_RealInverseFFTC(struct RealFFT* self,
+                              const int16_t* complex_data_in,
+                              int16_t* real_data_out) {
+  int i = 0;
+  int j = 0;
+  int result = 0;
+  int n = 1 << self->order;
+  // Create the buffer specific to complex-valued FFT implementation.
+  int16_t complex_buffer[2 << kMaxFFTOrder];
+
+  // For n-point FFT, first copy the first n + 2 elements into complex
+  // FFT, then construct the remaining n - 2 elements by real FFT's
+  // conjugate-symmetric properties.
+  memcpy(complex_buffer, complex_data_in, sizeof(int16_t) * (n + 2));
+  for (i = n + 2; i < 2 * n; i += 2) {
+    complex_buffer[i] = complex_data_in[2 * n - i];
+    complex_buffer[i + 1] = -complex_data_in[2 * n - i + 1];
+  }
+
+  WebRtcSpl_ComplexBitReverse(complex_buffer, self->order);
+  result = WebRtcSpl_ComplexIFFT(complex_buffer, self->order, 1);
+
+  // Strip out the imaginary parts of the complex inverse FFT output.
+  for (i = 0, j = 0; i < n; i += 1, j += 2) {
+    real_data_out[i] = complex_buffer[j];
+  }
+
+  return result;
+}
+
+#if defined(WEBRTC_DETECT_ARM_NEON) || defined(WEBRTC_ARCH_ARM_NEON)
+// TODO(kma): Replace the following function bodies into optimized functions
+// for ARM Neon.
+struct RealFFT* WebRtcSpl_CreateRealFFTNeon(int order) {
+  return WebRtcSpl_CreateRealFFTC(order);
+}
+
+void WebRtcSpl_FreeRealFFTNeon(struct RealFFT* self) {
+  WebRtcSpl_FreeRealFFTC(self);
+}
+
+int WebRtcSpl_RealForwardFFTNeon(struct RealFFT* self,
+                                 const int16_t* real_data_in,
+                                 int16_t* complex_data_out) {
+  return WebRtcSpl_RealForwardFFTC(self, real_data_in, complex_data_out);
+}
+
+int WebRtcSpl_RealInverseFFTNeon(struct RealFFT* self,
+                                 const int16_t* complex_data_in,
+                                 int16_t* real_data_out) {
+  return WebRtcSpl_RealInverseFFTC(self, complex_data_in, real_data_out);
+}
+#endif  // WEBRTC_DETECT_ARM_NEON || WEBRTC_ARCH_ARM_NEON
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample.c
new file mode 100644
index 0000000..b77738c
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample.c
@@ -0,0 +1,505 @@
+/*
+ *  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 resampling functions for 22 kHz.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "include/signal_processing_library.h"
+#include "resample_by_2_internal.h"
+
+// Declaration of internally used functions
+static void WebRtcSpl_32khzTo22khzIntToShort(const int32_t *In, int16_t *Out,
+                                             int32_t K);
+
+void WebRtcSpl_32khzTo22khzIntToInt(const int32_t *In, int32_t *Out,
+                                    int32_t K);
+
+// interpolation coefficients
+static const int16_t kCoefficients32To22[5][9] = {
+        {127, -712,  2359, -6333, 23456, 16775, -3695,  945, -154},
+        {-39,  230,  -830,  2785, 32366, -2324,   760, -218,   38},
+        {117, -663,  2222, -6133, 26634, 13070, -3174,  831, -137},
+        {-77,  457, -1677,  5958, 31175, -4136,  1405, -408,   71},
+        { 98, -560,  1900, -5406, 29240,  9423, -2480,  663, -110}
+};
+
+//////////////////////
+// 22 kHz -> 16 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 4, 5, 10
+#define SUB_BLOCKS_22_16    5
+
+// 22 -> 16 resampler
+void WebRtcSpl_Resample22khzTo16khz(const int16_t* in, int16_t* out,
+                                    WebRtcSpl_State22khzTo16khz* state, int32_t* tmpmem)
+{
+    int k;
+
+    // process two blocks of 10/SUB_BLOCKS_22_16 ms (to reduce temp buffer size)
+    for (k = 0; k < SUB_BLOCKS_22_16; k++)
+    {
+        ///// 22 --> 44 /////
+        // int16_t  in[220/SUB_BLOCKS_22_16]
+        // int32_t out[440/SUB_BLOCKS_22_16]
+        /////
+        WebRtcSpl_UpBy2ShortToInt(in, 220 / SUB_BLOCKS_22_16, tmpmem + 16, state->S_22_44);
+
+        ///// 44 --> 32 /////
+        // int32_t  in[440/SUB_BLOCKS_22_16]
+        // int32_t out[320/SUB_BLOCKS_22_16]
+        /////
+        // copy state to and from input array
+        tmpmem[8] = state->S_44_32[0];
+        tmpmem[9] = state->S_44_32[1];
+        tmpmem[10] = state->S_44_32[2];
+        tmpmem[11] = state->S_44_32[3];
+        tmpmem[12] = state->S_44_32[4];
+        tmpmem[13] = state->S_44_32[5];
+        tmpmem[14] = state->S_44_32[6];
+        tmpmem[15] = state->S_44_32[7];
+        state->S_44_32[0] = tmpmem[440 / SUB_BLOCKS_22_16 + 8];
+        state->S_44_32[1] = tmpmem[440 / SUB_BLOCKS_22_16 + 9];
+        state->S_44_32[2] = tmpmem[440 / SUB_BLOCKS_22_16 + 10];
+        state->S_44_32[3] = tmpmem[440 / SUB_BLOCKS_22_16 + 11];
+        state->S_44_32[4] = tmpmem[440 / SUB_BLOCKS_22_16 + 12];
+        state->S_44_32[5] = tmpmem[440 / SUB_BLOCKS_22_16 + 13];
+        state->S_44_32[6] = tmpmem[440 / SUB_BLOCKS_22_16 + 14];
+        state->S_44_32[7] = tmpmem[440 / SUB_BLOCKS_22_16 + 15];
+
+        WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 40 / SUB_BLOCKS_22_16);
+
+        ///// 32 --> 16 /////
+        // int32_t  in[320/SUB_BLOCKS_22_16]
+        // int32_t out[160/SUB_BLOCKS_22_16]
+        /////
+        WebRtcSpl_DownBy2IntToShort(tmpmem, 320 / SUB_BLOCKS_22_16, out, state->S_32_16);
+
+        // move input/output pointers 10/SUB_BLOCKS_22_16 ms seconds ahead
+        in += 220 / SUB_BLOCKS_22_16;
+        out += 160 / SUB_BLOCKS_22_16;
+    }
+}
+
+// initialize state of 22 -> 16 resampler
+void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state)
+{
+    int k;
+    for (k = 0; k < 8; k++)
+    {
+        state->S_22_44[k] = 0;
+        state->S_44_32[k] = 0;
+        state->S_32_16[k] = 0;
+    }
+}
+
+//////////////////////
+// 16 kHz -> 22 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 4, 5, 10
+#define SUB_BLOCKS_16_22    4
+
+// 16 -> 22 resampler
+void WebRtcSpl_Resample16khzTo22khz(const int16_t* in, int16_t* out,
+                                    WebRtcSpl_State16khzTo22khz* state, int32_t* tmpmem)
+{
+    int k;
+
+    // process two blocks of 10/SUB_BLOCKS_16_22 ms (to reduce temp buffer size)
+    for (k = 0; k < SUB_BLOCKS_16_22; k++)
+    {
+        ///// 16 --> 32 /////
+        // int16_t  in[160/SUB_BLOCKS_16_22]
+        // int32_t out[320/SUB_BLOCKS_16_22]
+        /////
+        WebRtcSpl_UpBy2ShortToInt(in, 160 / SUB_BLOCKS_16_22, tmpmem + 8, state->S_16_32);
+
+        ///// 32 --> 22 /////
+        // int32_t  in[320/SUB_BLOCKS_16_22]
+        // int32_t out[220/SUB_BLOCKS_16_22]
+        /////
+        // copy state to and from input array
+        tmpmem[0] = state->S_32_22[0];
+        tmpmem[1] = state->S_32_22[1];
+        tmpmem[2] = state->S_32_22[2];
+        tmpmem[3] = state->S_32_22[3];
+        tmpmem[4] = state->S_32_22[4];
+        tmpmem[5] = state->S_32_22[5];
+        tmpmem[6] = state->S_32_22[6];
+        tmpmem[7] = state->S_32_22[7];
+        state->S_32_22[0] = tmpmem[320 / SUB_BLOCKS_16_22];
+        state->S_32_22[1] = tmpmem[320 / SUB_BLOCKS_16_22 + 1];
+        state->S_32_22[2] = tmpmem[320 / SUB_BLOCKS_16_22 + 2];
+        state->S_32_22[3] = tmpmem[320 / SUB_BLOCKS_16_22 + 3];
+        state->S_32_22[4] = tmpmem[320 / SUB_BLOCKS_16_22 + 4];
+        state->S_32_22[5] = tmpmem[320 / SUB_BLOCKS_16_22 + 5];
+        state->S_32_22[6] = tmpmem[320 / SUB_BLOCKS_16_22 + 6];
+        state->S_32_22[7] = tmpmem[320 / SUB_BLOCKS_16_22 + 7];
+
+        WebRtcSpl_32khzTo22khzIntToShort(tmpmem, out, 20 / SUB_BLOCKS_16_22);
+
+        // move input/output pointers 10/SUB_BLOCKS_16_22 ms seconds ahead
+        in += 160 / SUB_BLOCKS_16_22;
+        out += 220 / SUB_BLOCKS_16_22;
+    }
+}
+
+// initialize state of 16 -> 22 resampler
+void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state)
+{
+    int k;
+    for (k = 0; k < 8; k++)
+    {
+        state->S_16_32[k] = 0;
+        state->S_32_22[k] = 0;
+    }
+}
+
+//////////////////////
+// 22 kHz ->  8 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 5, 10
+#define SUB_BLOCKS_22_8     2
+
+// 22 -> 8 resampler
+void WebRtcSpl_Resample22khzTo8khz(const int16_t* in, int16_t* out,
+                                   WebRtcSpl_State22khzTo8khz* state, int32_t* tmpmem)
+{
+    int k;
+
+    // process two blocks of 10/SUB_BLOCKS_22_8 ms (to reduce temp buffer size)
+    for (k = 0; k < SUB_BLOCKS_22_8; k++)
+    {
+        ///// 22 --> 22 lowpass /////
+        // int16_t  in[220/SUB_BLOCKS_22_8]
+        // int32_t out[220/SUB_BLOCKS_22_8]
+        /////
+        WebRtcSpl_LPBy2ShortToInt(in, 220 / SUB_BLOCKS_22_8, tmpmem + 16, state->S_22_22);
+
+        ///// 22 --> 16 /////
+        // int32_t  in[220/SUB_BLOCKS_22_8]
+        // int32_t out[160/SUB_BLOCKS_22_8]
+        /////
+        // copy state to and from input array
+        tmpmem[8] = state->S_22_16[0];
+        tmpmem[9] = state->S_22_16[1];
+        tmpmem[10] = state->S_22_16[2];
+        tmpmem[11] = state->S_22_16[3];
+        tmpmem[12] = state->S_22_16[4];
+        tmpmem[13] = state->S_22_16[5];
+        tmpmem[14] = state->S_22_16[6];
+        tmpmem[15] = state->S_22_16[7];
+        state->S_22_16[0] = tmpmem[220 / SUB_BLOCKS_22_8 + 8];
+        state->S_22_16[1] = tmpmem[220 / SUB_BLOCKS_22_8 + 9];
+        state->S_22_16[2] = tmpmem[220 / SUB_BLOCKS_22_8 + 10];
+        state->S_22_16[3] = tmpmem[220 / SUB_BLOCKS_22_8 + 11];
+        state->S_22_16[4] = tmpmem[220 / SUB_BLOCKS_22_8 + 12];
+        state->S_22_16[5] = tmpmem[220 / SUB_BLOCKS_22_8 + 13];
+        state->S_22_16[6] = tmpmem[220 / SUB_BLOCKS_22_8 + 14];
+        state->S_22_16[7] = tmpmem[220 / SUB_BLOCKS_22_8 + 15];
+
+        WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 20 / SUB_BLOCKS_22_8);
+
+        ///// 16 --> 8 /////
+        // int32_t in[160/SUB_BLOCKS_22_8]
+        // int32_t out[80/SUB_BLOCKS_22_8]
+        /////
+        WebRtcSpl_DownBy2IntToShort(tmpmem, 160 / SUB_BLOCKS_22_8, out, state->S_16_8);
+
+        // move input/output pointers 10/SUB_BLOCKS_22_8 ms seconds ahead
+        in += 220 / SUB_BLOCKS_22_8;
+        out += 80 / SUB_BLOCKS_22_8;
+    }
+}
+
+// initialize state of 22 -> 8 resampler
+void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state)
+{
+    int k;
+    for (k = 0; k < 8; k++)
+    {
+        state->S_22_22[k] = 0;
+        state->S_22_22[k + 8] = 0;
+        state->S_22_16[k] = 0;
+        state->S_16_8[k] = 0;
+    }
+}
+
+//////////////////////
+//  8 kHz -> 22 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 5, 10
+#define SUB_BLOCKS_8_22     2
+
+// 8 -> 22 resampler
+void WebRtcSpl_Resample8khzTo22khz(const int16_t* in, int16_t* out,
+                                   WebRtcSpl_State8khzTo22khz* state, int32_t* tmpmem)
+{
+    int k;
+
+    // process two blocks of 10/SUB_BLOCKS_8_22 ms (to reduce temp buffer size)
+    for (k = 0; k < SUB_BLOCKS_8_22; k++)
+    {
+        ///// 8 --> 16 /////
+        // int16_t  in[80/SUB_BLOCKS_8_22]
+        // int32_t out[160/SUB_BLOCKS_8_22]
+        /////
+        WebRtcSpl_UpBy2ShortToInt(in, 80 / SUB_BLOCKS_8_22, tmpmem + 18, state->S_8_16);
+
+        ///// 16 --> 11 /////
+        // int32_t  in[160/SUB_BLOCKS_8_22]
+        // int32_t out[110/SUB_BLOCKS_8_22]
+        /////
+        // copy state to and from input array
+        tmpmem[10] = state->S_16_11[0];
+        tmpmem[11] = state->S_16_11[1];
+        tmpmem[12] = state->S_16_11[2];
+        tmpmem[13] = state->S_16_11[3];
+        tmpmem[14] = state->S_16_11[4];
+        tmpmem[15] = state->S_16_11[5];
+        tmpmem[16] = state->S_16_11[6];
+        tmpmem[17] = state->S_16_11[7];
+        state->S_16_11[0] = tmpmem[160 / SUB_BLOCKS_8_22 + 10];
+        state->S_16_11[1] = tmpmem[160 / SUB_BLOCKS_8_22 + 11];
+        state->S_16_11[2] = tmpmem[160 / SUB_BLOCKS_8_22 + 12];
+        state->S_16_11[3] = tmpmem[160 / SUB_BLOCKS_8_22 + 13];
+        state->S_16_11[4] = tmpmem[160 / SUB_BLOCKS_8_22 + 14];
+        state->S_16_11[5] = tmpmem[160 / SUB_BLOCKS_8_22 + 15];
+        state->S_16_11[6] = tmpmem[160 / SUB_BLOCKS_8_22 + 16];
+        state->S_16_11[7] = tmpmem[160 / SUB_BLOCKS_8_22 + 17];
+
+        WebRtcSpl_32khzTo22khzIntToInt(tmpmem + 10, tmpmem, 10 / SUB_BLOCKS_8_22);
+
+        ///// 11 --> 22 /////
+        // int32_t  in[110/SUB_BLOCKS_8_22]
+        // int16_t out[220/SUB_BLOCKS_8_22]
+        /////
+        WebRtcSpl_UpBy2IntToShort(tmpmem, 110 / SUB_BLOCKS_8_22, out, state->S_11_22);
+
+        // move input/output pointers 10/SUB_BLOCKS_8_22 ms seconds ahead
+        in += 80 / SUB_BLOCKS_8_22;
+        out += 220 / SUB_BLOCKS_8_22;
+    }
+}
+
+// initialize state of 8 -> 22 resampler
+void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state)
+{
+    int k;
+    for (k = 0; k < 8; k++)
+    {
+        state->S_8_16[k] = 0;
+        state->S_16_11[k] = 0;
+        state->S_11_22[k] = 0;
+    }
+}
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_DotProdIntToInt(const int32_t* in1, const int32_t* in2,
+                                      const int16_t* coef_ptr, int32_t* out1,
+                                      int32_t* out2)
+{
+    int32_t tmp1 = 16384;
+    int32_t tmp2 = 16384;
+    int16_t coef;
+
+    coef = coef_ptr[0];
+    tmp1 += coef * in1[0];
+    tmp2 += coef * in2[-0];
+
+    coef = coef_ptr[1];
+    tmp1 += coef * in1[1];
+    tmp2 += coef * in2[-1];
+
+    coef = coef_ptr[2];
+    tmp1 += coef * in1[2];
+    tmp2 += coef * in2[-2];
+
+    coef = coef_ptr[3];
+    tmp1 += coef * in1[3];
+    tmp2 += coef * in2[-3];
+
+    coef = coef_ptr[4];
+    tmp1 += coef * in1[4];
+    tmp2 += coef * in2[-4];
+
+    coef = coef_ptr[5];
+    tmp1 += coef * in1[5];
+    tmp2 += coef * in2[-5];
+
+    coef = coef_ptr[6];
+    tmp1 += coef * in1[6];
+    tmp2 += coef * in2[-6];
+
+    coef = coef_ptr[7];
+    tmp1 += coef * in1[7];
+    tmp2 += coef * in2[-7];
+
+    coef = coef_ptr[8];
+    *out1 = tmp1 + coef * in1[8];
+    *out2 = tmp2 + coef * in2[-8];
+}
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_DotProdIntToShort(const int32_t* in1, const int32_t* in2,
+                                        const int16_t* coef_ptr, int16_t* out1,
+                                        int16_t* out2)
+{
+    int32_t tmp1 = 16384;
+    int32_t tmp2 = 16384;
+    int16_t coef;
+
+    coef = coef_ptr[0];
+    tmp1 += coef * in1[0];
+    tmp2 += coef * in2[-0];
+
+    coef = coef_ptr[1];
+    tmp1 += coef * in1[1];
+    tmp2 += coef * in2[-1];
+
+    coef = coef_ptr[2];
+    tmp1 += coef * in1[2];
+    tmp2 += coef * in2[-2];
+
+    coef = coef_ptr[3];
+    tmp1 += coef * in1[3];
+    tmp2 += coef * in2[-3];
+
+    coef = coef_ptr[4];
+    tmp1 += coef * in1[4];
+    tmp2 += coef * in2[-4];
+
+    coef = coef_ptr[5];
+    tmp1 += coef * in1[5];
+    tmp2 += coef * in2[-5];
+
+    coef = coef_ptr[6];
+    tmp1 += coef * in1[6];
+    tmp2 += coef * in2[-6];
+
+    coef = coef_ptr[7];
+    tmp1 += coef * in1[7];
+    tmp2 += coef * in2[-7];
+
+    coef = coef_ptr[8];
+    tmp1 += coef * in1[8];
+    tmp2 += coef * in2[-8];
+
+    // scale down, round and saturate
+    tmp1 >>= 15;
+    if (tmp1 > (int32_t)0x00007FFF)
+        tmp1 = 0x00007FFF;
+    if (tmp1 < (int32_t)0xFFFF8000)
+        tmp1 = 0xFFFF8000;
+    tmp2 >>= 15;
+    if (tmp2 > (int32_t)0x00007FFF)
+        tmp2 = 0x00007FFF;
+    if (tmp2 < (int32_t)0xFFFF8000)
+        tmp2 = 0xFFFF8000;
+    *out1 = (int16_t)tmp1;
+    *out2 = (int16_t)tmp2;
+}
+
+//   Resampling ratio: 11/16
+// input:  int32_t (normalized, not saturated) :: size 16 * K
+// output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 11 * K
+//      K: Number of blocks
+
+void WebRtcSpl_32khzTo22khzIntToInt(const int32_t* In,
+                                    int32_t* Out,
+                                    int32_t K)
+{
+    /////////////////////////////////////////////////////////////
+    // Filter operation:
+    //
+    // Perform resampling (16 input samples -> 11 output samples);
+    // process in sub blocks of size 16 samples.
+    int32_t m;
+
+    for (m = 0; m < K; m++)
+    {
+        // first output sample
+        Out[0] = ((int32_t)In[3] << 15) + (1 << 14);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToInt(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToInt(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToInt(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToInt(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToInt(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
+
+        // update pointers
+        In += 16;
+        Out += 11;
+    }
+}
+
+//   Resampling ratio: 11/16
+// input:  int32_t (normalized, not saturated) :: size 16 * K
+// output: int16_t (saturated) :: size 11 * K
+//      K: Number of blocks
+
+void WebRtcSpl_32khzTo22khzIntToShort(const int32_t *In,
+                                      int16_t *Out,
+                                      int32_t K)
+{
+    /////////////////////////////////////////////////////////////
+    // Filter operation:
+    //
+    // Perform resampling (16 input samples -> 11 output samples);
+    // process in sub blocks of size 16 samples.
+    int32_t tmp;
+    int32_t m;
+
+    for (m = 0; m < K; m++)
+    {
+        // first output sample
+        tmp = In[3];
+        if (tmp > (int32_t)0x00007FFF)
+            tmp = 0x00007FFF;
+        if (tmp < (int32_t)0xFFFF8000)
+            tmp = 0xFFFF8000;
+        Out[0] = (int16_t)tmp;
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToShort(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToShort(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToShort(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToShort(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToShort(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
+
+        // update pointers
+        In += 16;
+        Out += 11;
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_48khz.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_48khz.c
new file mode 100644
index 0000000..be55f45
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_48khz.c
@@ -0,0 +1,186 @@
+/*
+ *  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 resampling functions between 48 kHz and nb/wb.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include <string.h>
+#include "include/signal_processing_library.h"
+#include "resample_by_2_internal.h"
+
+////////////////////////////
+///// 48 kHz -> 16 kHz /////
+////////////////////////////
+
+// 48 -> 16 resampler
+void WebRtcSpl_Resample48khzTo16khz(const int16_t* in, int16_t* out,
+                                    WebRtcSpl_State48khzTo16khz* state, int32_t* tmpmem)
+{
+    ///// 48 --> 48(LP) /////
+    // int16_t  in[480]
+    // int32_t out[480]
+    /////
+    WebRtcSpl_LPBy2ShortToInt(in, 480, tmpmem + 16, state->S_48_48);
+
+    ///// 48 --> 32 /////
+    // int32_t  in[480]
+    // int32_t out[320]
+    /////
+    // copy state to and from input array
+    memcpy(tmpmem + 8, state->S_48_32, 8 * sizeof(int32_t));
+    memcpy(state->S_48_32, tmpmem + 488, 8 * sizeof(int32_t));
+    WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 160);
+
+    ///// 32 --> 16 /////
+    // int32_t  in[320]
+    // int16_t out[160]
+    /////
+    WebRtcSpl_DownBy2IntToShort(tmpmem, 320, out, state->S_32_16);
+}
+
+// initialize state of 48 -> 16 resampler
+void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state)
+{
+    memset(state->S_48_48, 0, 16 * sizeof(int32_t));
+    memset(state->S_48_32, 0, 8 * sizeof(int32_t));
+    memset(state->S_32_16, 0, 8 * sizeof(int32_t));
+}
+
+////////////////////////////
+///// 16 kHz -> 48 kHz /////
+////////////////////////////
+
+// 16 -> 48 resampler
+void WebRtcSpl_Resample16khzTo48khz(const int16_t* in, int16_t* out,
+                                    WebRtcSpl_State16khzTo48khz* state, int32_t* tmpmem)
+{
+    ///// 16 --> 32 /////
+    // int16_t  in[160]
+    // int32_t out[320]
+    /////
+    WebRtcSpl_UpBy2ShortToInt(in, 160, tmpmem + 16, state->S_16_32);
+
+    ///// 32 --> 24 /////
+    // int32_t  in[320]
+    // int32_t out[240]
+    // copy state to and from input array
+    /////
+    memcpy(tmpmem + 8, state->S_32_24, 8 * sizeof(int32_t));
+    memcpy(state->S_32_24, tmpmem + 328, 8 * sizeof(int32_t));
+    WebRtcSpl_Resample32khzTo24khz(tmpmem + 8, tmpmem, 80);
+
+    ///// 24 --> 48 /////
+    // int32_t  in[240]
+    // int16_t out[480]
+    /////
+    WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
+}
+
+// initialize state of 16 -> 48 resampler
+void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state)
+{
+    memset(state->S_16_32, 0, 8 * sizeof(int32_t));
+    memset(state->S_32_24, 0, 8 * sizeof(int32_t));
+    memset(state->S_24_48, 0, 8 * sizeof(int32_t));
+}
+
+////////////////////////////
+///// 48 kHz ->  8 kHz /////
+////////////////////////////
+
+// 48 -> 8 resampler
+void WebRtcSpl_Resample48khzTo8khz(const int16_t* in, int16_t* out,
+                                   WebRtcSpl_State48khzTo8khz* state, int32_t* tmpmem)
+{
+    ///// 48 --> 24 /////
+    // int16_t  in[480]
+    // int32_t out[240]
+    /////
+    WebRtcSpl_DownBy2ShortToInt(in, 480, tmpmem + 256, state->S_48_24);
+
+    ///// 24 --> 24(LP) /////
+    // int32_t  in[240]
+    // int32_t out[240]
+    /////
+    WebRtcSpl_LPBy2IntToInt(tmpmem + 256, 240, tmpmem + 16, state->S_24_24);
+
+    ///// 24 --> 16 /////
+    // int32_t  in[240]
+    // int32_t out[160]
+    /////
+    // copy state to and from input array
+    memcpy(tmpmem + 8, state->S_24_16, 8 * sizeof(int32_t));
+    memcpy(state->S_24_16, tmpmem + 248, 8 * sizeof(int32_t));
+    WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 80);
+
+    ///// 16 --> 8 /////
+    // int32_t  in[160]
+    // int16_t out[80]
+    /////
+    WebRtcSpl_DownBy2IntToShort(tmpmem, 160, out, state->S_16_8);
+}
+
+// initialize state of 48 -> 8 resampler
+void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state)
+{
+    memset(state->S_48_24, 0, 8 * sizeof(int32_t));
+    memset(state->S_24_24, 0, 16 * sizeof(int32_t));
+    memset(state->S_24_16, 0, 8 * sizeof(int32_t));
+    memset(state->S_16_8, 0, 8 * sizeof(int32_t));
+}
+
+////////////////////////////
+/////  8 kHz -> 48 kHz /////
+////////////////////////////
+
+// 8 -> 48 resampler
+void WebRtcSpl_Resample8khzTo48khz(const int16_t* in, int16_t* out,
+                                   WebRtcSpl_State8khzTo48khz* state, int32_t* tmpmem)
+{
+    ///// 8 --> 16 /////
+    // int16_t  in[80]
+    // int32_t out[160]
+    /////
+    WebRtcSpl_UpBy2ShortToInt(in, 80, tmpmem + 264, state->S_8_16);
+
+    ///// 16 --> 12 /////
+    // int32_t  in[160]
+    // int32_t out[120]
+    /////
+    // copy state to and from input array
+    memcpy(tmpmem + 256, state->S_16_12, 8 * sizeof(int32_t));
+    memcpy(state->S_16_12, tmpmem + 416, 8 * sizeof(int32_t));
+    WebRtcSpl_Resample32khzTo24khz(tmpmem + 256, tmpmem + 240, 40);
+
+    ///// 12 --> 24 /////
+    // int32_t  in[120]
+    // int16_t out[240]
+    /////
+    WebRtcSpl_UpBy2IntToInt(tmpmem + 240, 120, tmpmem, state->S_12_24);
+
+    ///// 24 --> 48 /////
+    // int32_t  in[240]
+    // int16_t out[480]
+    /////
+    WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
+}
+
+// initialize state of 8 -> 48 resampler
+void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state)
+{
+    memset(state->S_8_16, 0, 8 * sizeof(int32_t));
+    memset(state->S_16_12, 0, 8 * sizeof(int32_t));
+    memset(state->S_12_24, 0, 8 * sizeof(int32_t));
+    memset(state->S_24_48, 0, 8 * sizeof(int32_t));
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2.c
new file mode 100644
index 0000000..050ca5c
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2.c
@@ -0,0 +1,183 @@
+/*
+ *  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 resampling by two functions.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "include/signal_processing_library.h"
+
+#ifdef WEBRTC_ARCH_ARM_V7
+
+// allpass filter coefficients.
+static const uint32_t kResampleAllpass1[3] = {3284, 24441, 49528 << 15};
+static const uint32_t kResampleAllpass2[3] =
+  {12199, 37471 << 15, 60255 << 15};
+
+// Multiply two 32-bit values and accumulate to another input value.
+// Return: state + ((diff * tbl_value) >> 16)
+
+static __inline int32_t MUL_ACCUM_1(int32_t tbl_value,
+                                    int32_t diff,
+                                    int32_t state) {
+  int32_t result;
+  __asm __volatile ("smlawb %0, %1, %2, %3": "=r"(result): "r"(diff),
+                                   "r"(tbl_value), "r"(state));
+  return result;
+}
+
+// Multiply two 32-bit values and accumulate to another input value.
+// Return: Return: state + (((diff << 1) * tbl_value) >> 32)
+//
+// The reason to introduce this function is that, in case we can't use smlawb
+// instruction (in MUL_ACCUM_1) due to input value range, we can still use 
+// smmla to save some cycles.
+
+static __inline int32_t MUL_ACCUM_2(int32_t tbl_value,
+                                    int32_t diff,
+                                    int32_t state) {
+  int32_t result;
+  __asm __volatile ("smmla %0, %1, %2, %3": "=r"(result): "r"(diff << 1),
+                                  "r"(tbl_value), "r"(state));
+  return result;
+}
+
+#else
+
+// allpass filter coefficients.
+static const uint16_t kResampleAllpass1[3] = {3284, 24441, 49528};
+static const uint16_t kResampleAllpass2[3] = {12199, 37471, 60255};
+
+// Multiply a 32-bit value with a 16-bit value and accumulate to another input:
+#define MUL_ACCUM_1(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+#define MUL_ACCUM_2(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+
+#endif  // WEBRTC_ARCH_ARM_V7
+
+
+// decimator
+#if !defined(MIPS32_LE)
+void WebRtcSpl_DownsampleBy2(const int16_t* in, int16_t len,
+                             int16_t* out, int32_t* filtState) {
+  int32_t tmp1, tmp2, diff, in32, out32;
+  int16_t i;
+
+  register int32_t state0 = filtState[0];
+  register int32_t state1 = filtState[1];
+  register int32_t state2 = filtState[2];
+  register int32_t state3 = filtState[3];
+  register int32_t state4 = filtState[4];
+  register int32_t state5 = filtState[5];
+  register int32_t state6 = filtState[6];
+  register int32_t state7 = filtState[7];
+
+  for (i = (len >> 1); i > 0; i--) {
+    // lower allpass filter
+    in32 = (int32_t)(*in++) << 10;
+    diff = in32 - state1;
+    tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
+    state0 = in32;
+    diff = tmp1 - state2;
+    tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state1);
+    state1 = tmp1;
+    diff = tmp2 - state3;
+    state3 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state2);
+    state2 = tmp2;
+
+    // upper allpass filter
+    in32 = (int32_t)(*in++) << 10;
+    diff = in32 - state5;
+    tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
+    state4 = in32;
+    diff = tmp1 - state6;
+    tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state5);
+    state5 = tmp1;
+    diff = tmp2 - state7;
+    state7 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state6);
+    state6 = tmp2;
+
+    // add two allpass outputs, divide by two and round
+    out32 = (state3 + state7 + 1024) >> 11;
+
+    // limit amplitude to prevent wrap-around, and write to output array
+    *out++ = WebRtcSpl_SatW32ToW16(out32);
+  }
+
+  filtState[0] = state0;
+  filtState[1] = state1;
+  filtState[2] = state2;
+  filtState[3] = state3;
+  filtState[4] = state4;
+  filtState[5] = state5;
+  filtState[6] = state6;
+  filtState[7] = state7;
+}
+#endif  // #if defined(MIPS32_LE)
+
+
+void WebRtcSpl_UpsampleBy2(const int16_t* in, int16_t len,
+                           int16_t* out, int32_t* filtState) {
+  int32_t tmp1, tmp2, diff, in32, out32;
+  int16_t i;
+
+  register int32_t state0 = filtState[0];
+  register int32_t state1 = filtState[1];
+  register int32_t state2 = filtState[2];
+  register int32_t state3 = filtState[3];
+  register int32_t state4 = filtState[4];
+  register int32_t state5 = filtState[5];
+  register int32_t state6 = filtState[6];
+  register int32_t state7 = filtState[7];
+
+  for (i = len; i > 0; i--) {
+    // lower allpass filter
+    in32 = (int32_t)(*in++) << 10;
+    diff = in32 - state1;
+    tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state0);
+    state0 = in32;
+    diff = tmp1 - state2;
+    tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state1);
+    state1 = tmp1;
+    diff = tmp2 - state3;
+    state3 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state2);
+    state2 = tmp2;
+
+    // round; limit amplitude to prevent wrap-around; write to output array
+    out32 = (state3 + 512) >> 10;
+    *out++ = WebRtcSpl_SatW32ToW16(out32);
+
+    // upper allpass filter
+    diff = in32 - state5;
+    tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state4);
+    state4 = in32;
+    diff = tmp1 - state6;
+    tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state5);
+    state5 = tmp1;
+    diff = tmp2 - state7;
+    state7 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state6);
+    state6 = tmp2;
+
+    // round; limit amplitude to prevent wrap-around; write to output array
+    out32 = (state7 + 512) >> 10;
+    *out++ = WebRtcSpl_SatW32ToW16(out32);
+  }
+
+  filtState[0] = state0;
+  filtState[1] = state1;
+  filtState[2] = state2;
+  filtState[3] = state3;
+  filtState[4] = state4;
+  filtState[5] = state5;
+  filtState[6] = state6;
+  filtState[7] = state7;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2_internal.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2_internal.c
new file mode 100644
index 0000000..0f75957
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2_internal.c
@@ -0,0 +1,679 @@
+/*
+ *  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.
+ *
+ */
+
+#include "resample_by_2_internal.h"
+
+// allpass filter coefficients.
+static const int16_t kResampleAllpass[2][3] = {
+        {821, 6110, 12382},
+        {3050, 9368, 15063}
+};
+
+//
+//   decimator
+// input:  int32_t (shifted 15 positions to the left, + offset 16384) OVERWRITTEN!
+// output: int16_t (saturated) (of length len/2)
+// state:  filter state array; length = 8
+
+void WebRtcSpl_DownBy2IntToShort(int32_t *in, int32_t len, int16_t *out,
+                                 int32_t *state)
+{
+    int32_t tmp0, tmp1, diff;
+    int32_t i;
+
+    len >>= 1;
+
+    // lower allpass filter (operates on even input samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i << 1];
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // divide by two and store temporarily
+        in[i << 1] = (state[3] >> 1);
+    }
+
+    in++;
+
+    // upper allpass filter (operates on odd input samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i << 1];
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // divide by two and store temporarily
+        in[i << 1] = (state[7] >> 1);
+    }
+
+    in--;
+
+    // combine allpass outputs
+    for (i = 0; i < len; i += 2)
+    {
+        // divide by two, add both allpass outputs and round
+        tmp0 = (in[i << 1] + in[(i << 1) + 1]) >> 15;
+        tmp1 = (in[(i << 1) + 2] + in[(i << 1) + 3]) >> 15;
+        if (tmp0 > (int32_t)0x00007FFF)
+            tmp0 = 0x00007FFF;
+        if (tmp0 < (int32_t)0xFFFF8000)
+            tmp0 = 0xFFFF8000;
+        out[i] = (int16_t)tmp0;
+        if (tmp1 > (int32_t)0x00007FFF)
+            tmp1 = 0x00007FFF;
+        if (tmp1 < (int32_t)0xFFFF8000)
+            tmp1 = 0xFFFF8000;
+        out[i + 1] = (int16_t)tmp1;
+    }
+}
+
+//
+//   decimator
+// input:  int16_t
+// output: int32_t (shifted 15 positions to the left, + offset 16384) (of length len/2)
+// state:  filter state array; length = 8
+
+void WebRtcSpl_DownBy2ShortToInt(const int16_t *in,
+                                  int32_t len,
+                                  int32_t *out,
+                                  int32_t *state)
+{
+    int32_t tmp0, tmp1, diff;
+    int32_t i;
+
+    len >>= 1;
+
+    // lower allpass filter (operates on even input samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // divide by two and store temporarily
+        out[i] = (state[3] >> 1);
+    }
+
+    in++;
+
+    // upper allpass filter (operates on odd input samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // divide by two and store temporarily
+        out[i] += (state[7] >> 1);
+    }
+
+    in--;
+}
+
+//
+//   interpolator
+// input:  int16_t
+// output: int32_t (normalized, not saturated) (of length len*2)
+// state:  filter state array; length = 8
+void WebRtcSpl_UpBy2ShortToInt(const int16_t *in, int32_t len, int32_t *out,
+                               int32_t *state)
+{
+    int32_t tmp0, tmp1, diff;
+    int32_t i;
+
+    // upper allpass filter (generates odd output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((int32_t)in[i] << 15) + (1 << 14);
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[7] >> 15;
+    }
+
+    out++;
+
+    // lower allpass filter (generates even output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((int32_t)in[i] << 15) + (1 << 14);
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[3] >> 15;
+    }
+}
+
+//
+//   interpolator
+// input:  int32_t (shifted 15 positions to the left, + offset 16384)
+// output: int32_t (shifted 15 positions to the left, + offset 16384) (of length len*2)
+// state:  filter state array; length = 8
+void WebRtcSpl_UpBy2IntToInt(const int32_t *in, int32_t len, int32_t *out,
+                             int32_t *state)
+{
+    int32_t tmp0, tmp1, diff;
+    int32_t i;
+
+    // upper allpass filter (generates odd output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i];
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[7];
+    }
+
+    out++;
+
+    // lower allpass filter (generates even output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i];
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[3];
+    }
+}
+
+//
+//   interpolator
+// input:  int32_t (shifted 15 positions to the left, + offset 16384)
+// output: int16_t (saturated) (of length len*2)
+// state:  filter state array; length = 8
+void WebRtcSpl_UpBy2IntToShort(const int32_t *in, int32_t len, int16_t *out,
+                               int32_t *state)
+{
+    int32_t tmp0, tmp1, diff;
+    int32_t i;
+
+    // upper allpass filter (generates odd output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i];
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // scale down, saturate and store
+        tmp1 = state[7] >> 15;
+        if (tmp1 > (int32_t)0x00007FFF)
+            tmp1 = 0x00007FFF;
+        if (tmp1 < (int32_t)0xFFFF8000)
+            tmp1 = 0xFFFF8000;
+        out[i << 1] = (int16_t)tmp1;
+    }
+
+    out++;
+
+    // lower allpass filter (generates even output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i];
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // scale down, saturate and store
+        tmp1 = state[3] >> 15;
+        if (tmp1 > (int32_t)0x00007FFF)
+            tmp1 = 0x00007FFF;
+        if (tmp1 < (int32_t)0xFFFF8000)
+            tmp1 = 0xFFFF8000;
+        out[i << 1] = (int16_t)tmp1;
+    }
+}
+
+//   lowpass filter
+// input:  int16_t
+// output: int32_t (normalized, not saturated)
+// state:  filter state array; length = 8
+void WebRtcSpl_LPBy2ShortToInt(const int16_t* in, int32_t len, int32_t* out,
+                               int32_t* state)
+{
+    int32_t tmp0, tmp1, diff;
+    int32_t i;
+
+    len >>= 1;
+
+    // lower allpass filter: odd input -> even output samples
+    in++;
+    // initial state of polyphase delay element
+    tmp0 = state[12];
+    for (i = 0; i < len; i++)
+    {
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[3] >> 1;
+        tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+    }
+    in--;
+
+    // upper allpass filter: even input -> even output samples
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // average the two allpass outputs, scale down and store
+        out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
+    }
+
+    // switch to odd output samples
+    out++;
+
+    // lower allpass filter: even input -> odd output samples
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+        diff = tmp0 - state[9];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[8] + diff * kResampleAllpass[1][0];
+        state[8] = tmp0;
+        diff = tmp1 - state[10];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[9] + diff * kResampleAllpass[1][1];
+        state[9] = tmp1;
+        diff = tmp0 - state[11];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[11] = state[10] + diff * kResampleAllpass[1][2];
+        state[10] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[11] >> 1;
+    }
+
+    // upper allpass filter: odd input -> odd output samples
+    in++;
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+        diff = tmp0 - state[13];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[12] + diff * kResampleAllpass[0][0];
+        state[12] = tmp0;
+        diff = tmp1 - state[14];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[13] + diff * kResampleAllpass[0][1];
+        state[13] = tmp1;
+        diff = tmp0 - state[15];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[15] = state[14] + diff * kResampleAllpass[0][2];
+        state[14] = tmp0;
+
+        // average the two allpass outputs, scale down and store
+        out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
+    }
+}
+
+//   lowpass filter
+// input:  int32_t (shifted 15 positions to the left, + offset 16384)
+// output: int32_t (normalized, not saturated)
+// state:  filter state array; length = 8
+void WebRtcSpl_LPBy2IntToInt(const int32_t* in, int32_t len, int32_t* out,
+                             int32_t* state)
+{
+    int32_t tmp0, tmp1, diff;
+    int32_t i;
+
+    len >>= 1;
+
+    // lower allpass filter: odd input -> even output samples
+    in++;
+    // initial state of polyphase delay element
+    tmp0 = state[12];
+    for (i = 0; i < len; i++)
+    {
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[3] >> 1;
+        tmp0 = in[i << 1];
+    }
+    in--;
+
+    // upper allpass filter: even input -> even output samples
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i << 1];
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // average the two allpass outputs, scale down and store
+        out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
+    }
+
+    // switch to odd output samples
+    out++;
+
+    // lower allpass filter: even input -> odd output samples
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i << 1];
+        diff = tmp0 - state[9];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[8] + diff * kResampleAllpass[1][0];
+        state[8] = tmp0;
+        diff = tmp1 - state[10];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[9] + diff * kResampleAllpass[1][1];
+        state[9] = tmp1;
+        diff = tmp0 - state[11];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[11] = state[10] + diff * kResampleAllpass[1][2];
+        state[10] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[11] >> 1;
+    }
+
+    // upper allpass filter: odd input -> odd output samples
+    in++;
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i << 1];
+        diff = tmp0 - state[13];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[12] + diff * kResampleAllpass[0][0];
+        state[12] = tmp0;
+        diff = tmp1 - state[14];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[13] + diff * kResampleAllpass[0][1];
+        state[13] = tmp1;
+        diff = tmp0 - state[15];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[15] = state[14] + diff * kResampleAllpass[0][2];
+        state[14] = tmp0;
+
+        // average the two allpass outputs, scale down and store
+        out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2_internal.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2_internal.h
new file mode 100644
index 0000000..2819d56
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_by_2_internal.h
@@ -0,0 +1,47 @@
+/*
+ *  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 "../include/typedefs.h"
+
+/*******************************************************************
+ * resample_by_2_fast.c
+ * Functions for internal use in the other resample functions
+ ******************************************************************/
+void WebRtcSpl_DownBy2IntToShort(int32_t *in, int32_t len, int16_t *out,
+                                 int32_t *state);
+
+void WebRtcSpl_DownBy2ShortToInt(const int16_t *in, int32_t len,
+                                 int32_t *out, int32_t *state);
+
+void WebRtcSpl_UpBy2ShortToInt(const int16_t *in, int32_t len,
+                               int32_t *out, int32_t *state);
+
+void WebRtcSpl_UpBy2IntToInt(const int32_t *in, int32_t len, int32_t *out,
+                             int32_t *state);
+
+void WebRtcSpl_UpBy2IntToShort(const int32_t *in, int32_t len,
+                               int16_t *out, int32_t *state);
+
+void WebRtcSpl_LPBy2ShortToInt(const int16_t* in, int32_t len,
+                               int32_t* out, int32_t* state);
+
+void WebRtcSpl_LPBy2IntToInt(const int32_t* in, int32_t len, int32_t* out,
+                             int32_t* state);
+
+#endif // WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_fractional.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_fractional.c
new file mode 100644
index 0000000..4fb4c03
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/resample_fractional.c
@@ -0,0 +1,242 @@
+/*
+ *  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 resampling functions between 48, 44, 32 and 24 kHz.
+ * The description headers can be found in signal_processing_library.h
+ *
+ */
+
+#include "include/signal_processing_library.h"
+
+// interpolation coefficients
+static const int16_t kCoefficients48To32[2][8] = {
+        {778, -2050, 1087, 23285, 12903, -3783, 441, 222},
+        {222, 441, -3783, 12903, 23285, 1087, -2050, 778}
+};
+
+static const int16_t kCoefficients32To24[3][8] = {
+        {767, -2362, 2434, 24406, 10620, -3838, 721, 90},
+        {386, -381, -2646, 19062, 19062, -2646, -381, 386},
+        {90, 721, -3838, 10620, 24406, 2434, -2362, 767}
+};
+
+static const int16_t kCoefficients44To32[4][9] = {
+        {117, -669, 2245, -6183, 26267, 13529, -3245, 845, -138},
+        {-101, 612, -2283, 8532, 29790, -5138, 1789, -524, 91},
+        {50, -292, 1016, -3064, 32010, 3933, -1147, 315, -53},
+        {-156, 974, -3863, 18603, 21691, -6246, 2353, -712, 126}
+};
+
+//   Resampling ratio: 2/3
+// input:  int32_t (normalized, not saturated) :: size 3 * K
+// output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 2 * K
+//      K: number of blocks
+
+void WebRtcSpl_Resample48khzTo32khz(const int32_t *In, int32_t *Out,
+                                    int32_t K)
+{
+    /////////////////////////////////////////////////////////////
+    // Filter operation:
+    //
+    // Perform resampling (3 input samples -> 2 output samples);
+    // process in sub blocks of size 3 samples.
+    int32_t tmp;
+    int32_t m;
+
+    for (m = 0; m < K; m++)
+    {
+        tmp = 1 << 14;
+        tmp += kCoefficients48To32[0][0] * In[0];
+        tmp += kCoefficients48To32[0][1] * In[1];
+        tmp += kCoefficients48To32[0][2] * In[2];
+        tmp += kCoefficients48To32[0][3] * In[3];
+        tmp += kCoefficients48To32[0][4] * In[4];
+        tmp += kCoefficients48To32[0][5] * In[5];
+        tmp += kCoefficients48To32[0][6] * In[6];
+        tmp += kCoefficients48To32[0][7] * In[7];
+        Out[0] = tmp;
+
+        tmp = 1 << 14;
+        tmp += kCoefficients48To32[1][0] * In[1];
+        tmp += kCoefficients48To32[1][1] * In[2];
+        tmp += kCoefficients48To32[1][2] * In[3];
+        tmp += kCoefficients48To32[1][3] * In[4];
+        tmp += kCoefficients48To32[1][4] * In[5];
+        tmp += kCoefficients48To32[1][5] * In[6];
+        tmp += kCoefficients48To32[1][6] * In[7];
+        tmp += kCoefficients48To32[1][7] * In[8];
+        Out[1] = tmp;
+
+        // update pointers
+        In += 3;
+        Out += 2;
+    }
+}
+
+//   Resampling ratio: 3/4
+// input:  int32_t (normalized, not saturated) :: size 4 * K
+// output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 3 * K
+//      K: number of blocks
+
+void WebRtcSpl_Resample32khzTo24khz(const int32_t *In, int32_t *Out,
+                                    int32_t K)
+{
+    /////////////////////////////////////////////////////////////
+    // Filter operation:
+    //
+    // Perform resampling (4 input samples -> 3 output samples);
+    // process in sub blocks of size 4 samples.
+    int32_t m;
+    int32_t tmp;
+
+    for (m = 0; m < K; m++)
+    {
+        tmp = 1 << 14;
+        tmp += kCoefficients32To24[0][0] * In[0];
+        tmp += kCoefficients32To24[0][1] * In[1];
+        tmp += kCoefficients32To24[0][2] * In[2];
+        tmp += kCoefficients32To24[0][3] * In[3];
+        tmp += kCoefficients32To24[0][4] * In[4];
+        tmp += kCoefficients32To24[0][5] * In[5];
+        tmp += kCoefficients32To24[0][6] * In[6];
+        tmp += kCoefficients32To24[0][7] * In[7];
+        Out[0] = tmp;
+
+        tmp = 1 << 14;
+        tmp += kCoefficients32To24[1][0] * In[1];
+        tmp += kCoefficients32To24[1][1] * In[2];
+        tmp += kCoefficients32To24[1][2] * In[3];
+        tmp += kCoefficients32To24[1][3] * In[4];
+        tmp += kCoefficients32To24[1][4] * In[5];
+        tmp += kCoefficients32To24[1][5] * In[6];
+        tmp += kCoefficients32To24[1][6] * In[7];
+        tmp += kCoefficients32To24[1][7] * In[8];
+        Out[1] = tmp;
+
+        tmp = 1 << 14;
+        tmp += kCoefficients32To24[2][0] * In[2];
+        tmp += kCoefficients32To24[2][1] * In[3];
+        tmp += kCoefficients32To24[2][2] * In[4];
+        tmp += kCoefficients32To24[2][3] * In[5];
+        tmp += kCoefficients32To24[2][4] * In[6];
+        tmp += kCoefficients32To24[2][5] * In[7];
+        tmp += kCoefficients32To24[2][6] * In[8];
+        tmp += kCoefficients32To24[2][7] * In[9];
+        Out[2] = tmp;
+
+        // update pointers
+        In += 4;
+        Out += 3;
+    }
+}
+
+//
+// fractional resampling filters
+//   Fout = 11/16 * Fin
+//   Fout =  8/11 * Fin
+//
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_ResampDotProduct(const int32_t *in1, const int32_t *in2,
+                                       const int16_t *coef_ptr, int32_t *out1,
+                                       int32_t *out2)
+{
+    int32_t tmp1 = 16384;
+    int32_t tmp2 = 16384;
+    int16_t coef;
+
+    coef = coef_ptr[0];
+    tmp1 += coef * in1[0];
+    tmp2 += coef * in2[-0];
+
+    coef = coef_ptr[1];
+    tmp1 += coef * in1[1];
+    tmp2 += coef * in2[-1];
+
+    coef = coef_ptr[2];
+    tmp1 += coef * in1[2];
+    tmp2 += coef * in2[-2];
+
+    coef = coef_ptr[3];
+    tmp1 += coef * in1[3];
+    tmp2 += coef * in2[-3];
+
+    coef = coef_ptr[4];
+    tmp1 += coef * in1[4];
+    tmp2 += coef * in2[-4];
+
+    coef = coef_ptr[5];
+    tmp1 += coef * in1[5];
+    tmp2 += coef * in2[-5];
+
+    coef = coef_ptr[6];
+    tmp1 += coef * in1[6];
+    tmp2 += coef * in2[-6];
+
+    coef = coef_ptr[7];
+    tmp1 += coef * in1[7];
+    tmp2 += coef * in2[-7];
+
+    coef = coef_ptr[8];
+    *out1 = tmp1 + coef * in1[8];
+    *out2 = tmp2 + coef * in2[-8];
+}
+
+//   Resampling ratio: 8/11
+// input:  int32_t (normalized, not saturated) :: size 11 * K
+// output: int32_t (shifted 15 positions to the left, + offset 16384) :: size  8 * K
+//      K: number of blocks
+
+void WebRtcSpl_Resample44khzTo32khz(const int32_t *In, int32_t *Out,
+                                    int32_t K)
+{
+    /////////////////////////////////////////////////////////////
+    // Filter operation:
+    //
+    // Perform resampling (11 input samples -> 8 output samples);
+    // process in sub blocks of size 11 samples.
+    int32_t tmp;
+    int32_t m;
+
+    for (m = 0; m < K; m++)
+    {
+        tmp = 1 << 14;
+
+        // first output sample
+        Out[0] = ((int32_t)In[3] << 15) + tmp;
+
+        // sum and accumulate filter coefficients and input samples
+        tmp += kCoefficients44To32[3][0] * In[5];
+        tmp += kCoefficients44To32[3][1] * In[6];
+        tmp += kCoefficients44To32[3][2] * In[7];
+        tmp += kCoefficients44To32[3][3] * In[8];
+        tmp += kCoefficients44To32[3][4] * In[9];
+        tmp += kCoefficients44To32[3][5] * In[10];
+        tmp += kCoefficients44To32[3][6] * In[11];
+        tmp += kCoefficients44To32[3][7] * In[12];
+        tmp += kCoefficients44To32[3][8] * In[13];
+        Out[4] = tmp;
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_ResampDotProduct(&In[0], &In[17], kCoefficients44To32[0], &Out[1], &Out[7]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_ResampDotProduct(&In[2], &In[15], kCoefficients44To32[1], &Out[2], &Out[6]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_ResampDotProduct(&In[3], &In[14], kCoefficients44To32[2], &Out[3], &Out[5]);
+
+        // update pointers
+        In += 11;
+        Out += 8;
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_init.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_init.c
new file mode 100644
index 0000000..9e0e8b0
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_init.c
@@ -0,0 +1,157 @@
+/*
+ *  Copyright (c) 2012 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 global function contained in this file initializes SPL function
+ * pointers, currently only for ARM platforms.
+ *
+ * Some code came from common/rtcd.c in the WebM project.
+ */
+
+#include "include/real_fft.h"
+#include "include/signal_processing_library.h"
+
+
+/* Declare function pointers. */
+MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16;
+MaxAbsValueW32 WebRtcSpl_MaxAbsValueW32;
+MaxValueW16 WebRtcSpl_MaxValueW16;
+MaxValueW32 WebRtcSpl_MaxValueW32;
+MinValueW16 WebRtcSpl_MinValueW16;
+MinValueW32 WebRtcSpl_MinValueW32;
+CrossCorrelation WebRtcSpl_CrossCorrelation;
+DownsampleFast WebRtcSpl_DownsampleFast;
+ScaleAndAddVectorsWithRound WebRtcSpl_ScaleAndAddVectorsWithRound;
+CreateRealFFT WebRtcSpl_CreateRealFFT;
+FreeRealFFT WebRtcSpl_FreeRealFFT;
+RealForwardFFT WebRtcSpl_RealForwardFFT;
+RealInverseFFT WebRtcSpl_RealInverseFFT;
+
+#if (defined(WEBRTC_DETECT_ARM_NEON) || !defined(WEBRTC_ARCH_ARM_NEON)) && \
+     !defined(MIPS32_LE)
+/* Initialize function pointers to the generic C version. */
+static void InitPointersToC() {
+  WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16C;
+  WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32C;
+  WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16C;
+  WebRtcSpl_MaxValueW32 = WebRtcSpl_MaxValueW32C;
+  WebRtcSpl_MinValueW16 = WebRtcSpl_MinValueW16C;
+  WebRtcSpl_MinValueW32 = WebRtcSpl_MinValueW32C;
+  WebRtcSpl_CrossCorrelation = WebRtcSpl_CrossCorrelationC;
+  WebRtcSpl_DownsampleFast = WebRtcSpl_DownsampleFastC;
+  WebRtcSpl_ScaleAndAddVectorsWithRound =
+      WebRtcSpl_ScaleAndAddVectorsWithRoundC;
+  WebRtcSpl_CreateRealFFT = WebRtcSpl_CreateRealFFTC;
+  WebRtcSpl_FreeRealFFT = WebRtcSpl_FreeRealFFTC;
+  WebRtcSpl_RealForwardFFT = WebRtcSpl_RealForwardFFTC;
+  WebRtcSpl_RealInverseFFT = WebRtcSpl_RealInverseFFTC;
+}
+#endif
+
+#if defined(WEBRTC_DETECT_ARM_NEON) || defined(WEBRTC_ARCH_ARM_NEON)
+/* Initialize function pointers to the Neon version. */
+static void InitPointersToNeon() {
+  WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16Neon;
+  WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32Neon;
+  WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16Neon;
+  WebRtcSpl_MaxValueW32 = WebRtcSpl_MaxValueW32Neon;
+  WebRtcSpl_MinValueW16 = WebRtcSpl_MinValueW16Neon;
+  WebRtcSpl_MinValueW32 = WebRtcSpl_MinValueW32Neon;
+  WebRtcSpl_CrossCorrelation = WebRtcSpl_CrossCorrelationNeon;
+  WebRtcSpl_DownsampleFast = WebRtcSpl_DownsampleFastNeon;
+  WebRtcSpl_ScaleAndAddVectorsWithRound =
+      WebRtcSpl_ScaleAndAddVectorsWithRoundNeon;
+  WebRtcSpl_CreateRealFFT = WebRtcSpl_CreateRealFFTNeon;
+  WebRtcSpl_FreeRealFFT = WebRtcSpl_FreeRealFFTNeon;
+  WebRtcSpl_RealForwardFFT = WebRtcSpl_RealForwardFFTNeon;
+  WebRtcSpl_RealInverseFFT = WebRtcSpl_RealInverseFFTNeon;
+}
+#endif
+
+#if defined(MIPS32_LE)
+/* Initialize function pointers to the MIPS version. */
+static void InitPointersToMIPS() {
+  WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16_mips;
+  WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16_mips;
+  WebRtcSpl_MaxValueW32 = WebRtcSpl_MaxValueW32_mips;
+  WebRtcSpl_MinValueW16 = WebRtcSpl_MinValueW16_mips;
+  WebRtcSpl_MinValueW32 = WebRtcSpl_MinValueW32_mips;
+  WebRtcSpl_CrossCorrelation = WebRtcSpl_CrossCorrelation_mips;
+  WebRtcSpl_DownsampleFast = WebRtcSpl_DownsampleFast_mips;
+  WebRtcSpl_CreateRealFFT = WebRtcSpl_CreateRealFFTC;
+  WebRtcSpl_FreeRealFFT = WebRtcSpl_FreeRealFFTC;
+  WebRtcSpl_RealForwardFFT = WebRtcSpl_RealForwardFFTC;
+  WebRtcSpl_RealInverseFFT = WebRtcSpl_RealInverseFFTC;
+#if defined(MIPS_DSP_R1_LE)
+  WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32_mips;
+  WebRtcSpl_ScaleAndAddVectorsWithRound =
+      WebRtcSpl_ScaleAndAddVectorsWithRound_mips;
+#else
+  WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32C;
+  WebRtcSpl_ScaleAndAddVectorsWithRound =
+      WebRtcSpl_ScaleAndAddVectorsWithRoundC;
+#endif
+}
+#endif
+
+static void InitFunctionPointers(void) {
+#if defined(WEBRTC_DETECT_ARM_NEON)
+  if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
+    InitPointersToNeon();
+  } else {
+    InitPointersToC();
+  }
+#elif defined(WEBRTC_ARCH_ARM_NEON)
+  InitPointersToNeon();
+#elif defined(MIPS32_LE)
+  InitPointersToMIPS();
+#else
+  InitPointersToC();
+#endif  /* WEBRTC_DETECT_ARM_NEON */
+}
+
+#if defined(WEBRTC_POSIX)
+#include <pthread.h>
+
+static void once(void (*func)(void)) {
+  static pthread_once_t lock = PTHREAD_ONCE_INIT;
+  pthread_once(&lock, func);
+}
+
+#elif defined(_WIN32)
+#include <windows.h>
+
+static void once(void (*func)(void)) {
+  /* Didn't use InitializeCriticalSection() since there's no race-free context
+   * in which to execute it.
+   *
+   * TODO(kma): Change to different implementation (e.g.
+   * InterlockedCompareExchangePointer) to avoid issues similar to
+   * http://code.google.com/p/webm/issues/detail?id=467.
+   */
+  static CRITICAL_SECTION lock = {(void *)((size_t)-1), -1, 0, 0, 0, 0};
+  static int done = 0;
+
+  EnterCriticalSection(&lock);
+  if (!done) {
+    func();
+    done = 1;
+  }
+  LeaveCriticalSection(&lock);
+}
+
+/* There's no fallback version as an #else block here to ensure thread safety.
+ * In case of neither pthread for WEBRTC_POSIX nor _WIN32 is present, build
+ * system should pick it up.
+ */
+#endif  /* WEBRTC_POSIX */
+
+void WebRtcSpl_Init() {
+  once(InitFunctionPointers);
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_sqrt.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_sqrt.c
new file mode 100644
index 0000000..d6540fd
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_sqrt.c
@@ -0,0 +1,184 @@
+/*
+ *  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_Sqrt().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "include/signal_processing_library.h"
+
+int32_t WebRtcSpl_SqrtLocal(int32_t in);
+
+int32_t WebRtcSpl_SqrtLocal(int32_t in)
+{
+
+    int16_t x_half, t16;
+    int32_t A, B, x2;
+
+    /* The following block performs:
+     y=in/2
+     x=y-2^30
+     x_half=x/2^31
+     t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
+         + 0.875*((x_half)^5)
+     */
+
+    B = in;
+
+    B = WEBRTC_SPL_RSHIFT_W32(B, 1); // B = in/2
+    B = B - ((int32_t)0x40000000); // B = in/2 - 1/2
+    x_half = (int16_t)WEBRTC_SPL_RSHIFT_W32(B, 16);// x_half = x/2 = (in-1)/2
+    B = B + ((int32_t)0x40000000); // B = 1 + x/2
+    B = B + ((int32_t)0x40000000); // Add 0.5 twice (since 1.0 does not exist in Q31)
+
+    x2 = ((int32_t)x_half) * ((int32_t)x_half) * 2; // A = (x/2)^2
+    A = -x2; // A = -(x/2)^2
+    B = B + (A >> 1); // B = 1 + x/2 - 0.5*(x/2)^2
+
+    A = WEBRTC_SPL_RSHIFT_W32(A, 16);
+    A = A * A * 2; // A = (x/2)^4
+    t16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16);
+    B = B + WEBRTC_SPL_MUL_16_16(-20480, t16) * 2; // B = B - 0.625*A
+    // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4
+
+    t16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16);
+    A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = (x/2)^5
+    t16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16);
+    B = B + WEBRTC_SPL_MUL_16_16(28672, t16) * 2; // B = B + 0.875*A
+    // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4 + 0.875*(x/2)^5
+
+    t16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(x2, 16);
+    A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = x/2^3
+
+    B = B + (A >> 1); // B = B + 0.5*A
+    // After this, B = 1 + x/2 - 0.5*(x/2)^2 + 0.5*(x/2)^3 - 0.625*(x/2)^4 + 0.875*(x/2)^5
+
+    B = B + ((int32_t)32768); // Round off bit
+
+    return B;
+}
+
+int32_t WebRtcSpl_Sqrt(int32_t value)
+{
+    /*
+     Algorithm:
+
+     Six term Taylor Series is used here to compute the square root of a number
+     y^0.5 = (1+x)^0.5 where x = y-1
+     = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
+     0.5 <= x < 1
+
+     Example of how the algorithm works, with ut=sqrt(in), and
+     with in=73632 and ut=271 (even shift value case):
+
+     in=73632
+     y= in/131072
+     x=y-1
+     t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
+     ut=t*(1/sqrt(2))*512
+
+     or:
+
+     in=73632
+     in2=73632*2^14
+     y= in2/2^31
+     x=y-1
+     t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
+     ut=t*(1/sqrt(2))
+     ut2=ut*2^9
+
+     which gives:
+
+     in  = 73632
+     in2 = 1206386688
+     y   = 0.56176757812500
+     x   = -0.43823242187500
+     t   = 0.74973506527313
+     ut  = 0.53014274874797
+     ut2 = 2.714330873589594e+002
+
+     or:
+
+     in=73632
+     in2=73632*2^14
+     y=in2/2
+     x=y-2^30
+     x_half=x/2^31
+     t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
+         + 0.875*((x_half)^5)
+     ut=t*(1/sqrt(2))
+     ut2=ut*2^9
+
+     which gives:
+
+     in  = 73632
+     in2 = 1206386688
+     y   = 603193344
+     x   = -470548480
+     x_half =  -0.21911621093750
+     t   = 0.74973506527313
+     ut  = 0.53014274874797
+     ut2 = 2.714330873589594e+002
+
+     */
+
+    int16_t x_norm, nshift, t16, sh;
+    int32_t A;
+
+    int16_t k_sqrt_2 = 23170; // 1/sqrt2 (==5a82)
+
+    A = value;
+
+    if (A == 0)
+        return (int32_t)0; // sqrt(0) = 0
+
+    sh = WebRtcSpl_NormW32(A); // # shifts to normalize A
+    A = WEBRTC_SPL_LSHIFT_W32(A, sh); // Normalize A
+    if (A < (WEBRTC_SPL_WORD32_MAX - 32767))
+    {
+        A = A + ((int32_t)32768); // Round off bit
+    } else
+    {
+        A = WEBRTC_SPL_WORD32_MAX;
+    }
+
+    x_norm = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16); // x_norm = AH
+
+    nshift = WEBRTC_SPL_RSHIFT_W16(sh, 1); // nshift = sh>>1
+    nshift = -nshift; // Negate the power for later de-normalization
+
+    A = (int32_t)WEBRTC_SPL_LSHIFT_W32((int32_t)x_norm, 16);
+    A = WEBRTC_SPL_ABS_W32(A); // A = abs(x_norm<<16)
+    A = WebRtcSpl_SqrtLocal(A); // A = sqrt(A)
+
+    if ((-2 * nshift) == sh)
+    { // Even shift value case
+
+        t16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(A, 16); // t16 = AH
+
+        A = WEBRTC_SPL_MUL_16_16(k_sqrt_2, t16) * 2; // A = 1/sqrt(2)*t16
+        A = A + ((int32_t)32768); // Round off
+        A = A & ((int32_t)0x7fff0000); // Round off
+
+        A = WEBRTC_SPL_RSHIFT_W32(A, 15); // A = A>>16
+
+    } else
+    {
+        A = WEBRTC_SPL_RSHIFT_W32(A, 16); // A = A>>16
+    }
+
+    A = A & ((int32_t)0x0000ffff);
+    A = (int32_t)WEBRTC_SPL_SHIFT_W32(A, nshift); // De-normalize the result
+
+    return A;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_version.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_version.c
new file mode 100644
index 0000000..20a4b4c
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/spl_version.c
@@ -0,0 +1,25 @@
+/*
+ *  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 "include/signal_processing_library.h"
+
+int16_t WebRtcSpl_get_version(char* version, int16_t length_in_bytes)
+{
+    strncpy(version, "1.2.0", length_in_bytes);
+    return 0;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
new file mode 100644
index 0000000..8352c76
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
@@ -0,0 +1,35 @@
+/*
+ *  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_SqrtOfOneMinusXSquared().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "include/signal_processing_library.h"
+
+void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t *xQ15, int vector_length,
+                                      int16_t *yQ15)
+{
+    int32_t sq;
+    int m;
+    int16_t tmp;
+
+    for (m = 0; m < vector_length; m++)
+    {
+        tmp = xQ15[m];
+        sq = WEBRTC_SPL_MUL_16_16(tmp, tmp); // x^2 in Q30
+        sq = 1073741823 - sq; // 1-x^2, where 1 ~= 0.99999999906 is 1073741823 in Q30
+        sq = WebRtcSpl_Sqrt(sq); // sqrt(1-x^2) in Q15
+        yQ15[m] = (int16_t)sq;
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/vector_scaling_operations.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/vector_scaling_operations.c
new file mode 100644
index 0000000..214bab1
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/signal_processing/vector_scaling_operations.c
@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2012 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()
+ * WebRtcSpl_ScaleAndAddVectorsWithRoundC()
+ */
+
+#include "include/signal_processing_library.h"
+
+void WebRtcSpl_VectorBitShiftW16(int16_t *res, int16_t length,
+                                 const int16_t *in, int16_t 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(int32_t *out_vector,
+                                 int16_t vector_length,
+                                 const int32_t *in_vector,
+                                 int16_t 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(int16_t* out, int length,
+                                      const int32_t* in, int right_shifts) {
+  int i;
+  int32_t tmp_w32;
+
+  if (right_shifts >= 0) {
+    for (i = length; i > 0; i--) {
+      tmp_w32 = (*in++) >> right_shifts;
+      (*out++) = WebRtcSpl_SatW32ToW16(tmp_w32);
+    }
+  } else {
+    int16_t left_shifts = -right_shifts;
+    for (i = length; i > 0; i--) {
+      tmp_w32 = (*in++) << left_shifts;
+      (*out++) = WebRtcSpl_SatW32ToW16(tmp_w32);
+    }
+  }
+}
+
+void WebRtcSpl_ScaleVector(const int16_t *in_vector, int16_t *out_vector,
+                           int16_t gain, int16_t in_vector_length,
+                           int16_t right_shifts)
+{
+    // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
+    int i;
+    const int16_t *inptr;
+    int16_t *outptr;
+
+    inptr = in_vector;
+    outptr = out_vector;
+
+    for (i = 0; i < in_vector_length; i++)
+    {
+        (*outptr++) = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
+    }
+}
+
+void WebRtcSpl_ScaleVectorWithSat(const int16_t *in_vector, int16_t *out_vector,
+                                 int16_t gain, int16_t in_vector_length,
+                                 int16_t right_shifts)
+{
+    // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
+    int i;
+    int32_t tmpW32;
+    const int16_t *inptr;
+    int16_t *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(const int16_t *in1, int16_t gain1, int shift1,
+                                  const int16_t *in2, int16_t gain2, int shift2,
+                                  int16_t *out, int vector_length)
+{
+    // Performs vector operation: out = (gain1*in1)>>shift1 + (gain2*in2)>>shift2
+    int i;
+    const int16_t *in1ptr;
+    const int16_t *in2ptr;
+    int16_t *outptr;
+
+    in1ptr = in1;
+    in2ptr = in2;
+    outptr = out;
+
+    for (i = 0; i < vector_length; i++)
+    {
+        (*outptr++) = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(gain1, *in1ptr++, shift1)
+                + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(gain2, *in2ptr++, shift2);
+    }
+}
+
+// C version of WebRtcSpl_ScaleAndAddVectorsWithRound() for generic platforms.
+int WebRtcSpl_ScaleAndAddVectorsWithRoundC(const int16_t* in_vector1,
+                                           int16_t in_vector1_scale,
+                                           const int16_t* in_vector2,
+                                           int16_t in_vector2_scale,
+                                           int right_shifts,
+                                           int16_t* out_vector,
+                                           int length) {
+  int i = 0;
+  int round_value = (1 << right_shifts) >> 1;
+
+  if (in_vector1 == NULL || in_vector2 == NULL || out_vector == NULL ||
+      length <= 0 || right_shifts < 0) {
+    return -1;
+  }
+
+  for (i = 0; i < length; i++) {
+    out_vector[i] = (int16_t)((
+        WEBRTC_SPL_MUL_16_16(in_vector1[i], in_vector1_scale)
+        + WEBRTC_SPL_MUL_16_16(in_vector2[i], in_vector2_scale)
+        + round_value) >> right_shifts);
+  }
+
+  return 0;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/Android.mk b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/Android.mk
new file mode 100644
index 0000000..ed5de6b
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/Android.mk
@@ -0,0 +1,51 @@
+# 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.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_vad
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+    webrtc_vad.c \
+    vad_core.c \
+    vad_filterbank.c \
+    vad_gmm.c \
+    vad_sp.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+    $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+    $(LOCAL_PATH)/include \
+    $(LOCAL_PATH)/../.. \
+    $(LOCAL_PATH)/../signal_processing/include \
+    external/webrtc
+
+LOCAL_SHARED_LIBRARIES := \
+    libdl \
+    libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/include/webrtc_vad.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/include/webrtc_vad.h
new file mode 100644
index 0000000..88041a7
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/include/webrtc_vad.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2012 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_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_  // NOLINT
+#define WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
+
+#include "../../include/typedefs.h"
+
+
+typedef struct WebRtcVadInst VadInst;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+// Creates an instance to the VAD structure.
+//
+// - handle [o] : Pointer to the VAD instance that should be created.
+//
+// returns      : 0 - (OK), -1 - (Error)
+int WebRtcVad_Create(VadInst** handle);
+
+// Frees the dynamic memory of a specified VAD instance.
+//
+// - handle [i] : Pointer to VAD instance that should be freed.
+//
+// returns      : 0 - (OK), -1 - (NULL pointer in)
+int WebRtcVad_Free(VadInst* handle);
+
+// Initializes a VAD instance.
+//
+// - handle [i/o] : Instance that should be initialized.
+//
+// returns        : 0 - (OK),
+//                 -1 - (NULL pointer or Default mode could not be set).
+int WebRtcVad_Init(VadInst* handle);
+
+// Sets the VAD operating mode. A more aggressive (higher mode) VAD is more
+// restrictive in reporting speech. Put in other words the probability of being
+// speech when the VAD returns 1 is increased with increasing mode. As a
+// consequence also the missed detection rate goes up.
+//
+// - handle [i/o] : VAD instance.
+// - mode   [i]   : Aggressiveness mode (0, 1, 2, or 3).
+//
+// returns        : 0 - (OK),
+//                 -1 - (NULL pointer, mode could not be set or the VAD instance
+//                       has not been initialized).
+int WebRtcVad_set_mode(VadInst* handle, int mode);
+
+// Calculates a VAD decision for the |audio_frame|. For valid sampling rates
+// frame lengths, see the description of WebRtcVad_ValidRatesAndFrameLengths().
+//
+// - handle       [i/o] : VAD Instance. Needs to be initialized by
+//                        WebRtcVad_Init() before call.
+// - fs           [i]   : Sampling frequency (Hz): 8000, 16000, or 32000
+// - audio_frame  [i]   : Audio frame buffer.
+// - frame_length [i]   : Length of audio frame buffer in number of samples.
+//
+// returns              : 1 - (Active Voice),
+//                        0 - (Non-active Voice),
+//                       -1 - (Error)
+int WebRtcVad_Process(VadInst* handle, int fs, int16_t* audio_frame,
+                      int frame_length);
+
+// Checks for valid combinations of |rate| and |frame_length|. We support 10,
+// 20 and 30 ms frames and the rates 8000, 16000 and 32000 Hz.
+//
+// - rate         [i] : Sampling frequency (Hz).
+// - frame_length [i] : Speech frame buffer length in number of samples.
+//
+// returns            : 0 - (valid combination), -1 - (invalid combination)
+int WebRtcVad_ValidRateAndFrameLength(int rate, int frame_length);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_  // NOLINT
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_core.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_core.c
new file mode 100644
index 0000000..81a2119
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_core.c
@@ -0,0 +1,682 @@
+/*
+ *  Copyright (c) 2012 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 "vad_core.h"
+
+#include "../signal_processing/include/signal_processing_library.h"
+#include "vad_filterbank.h"
+#include "vad_gmm.h"
+#include "vad_sp.h"
+#include "../include/typedefs.h"
+
+// Spectrum Weighting
+static const int16_t kSpectrumWeight[kNumChannels] = { 6, 8, 10, 12, 14, 16 };
+static const int16_t kNoiseUpdateConst = 655; // Q15
+static const int16_t kSpeechUpdateConst = 6554; // Q15
+static const int16_t kBackEta = 154; // Q8
+// Minimum difference between the two models, Q5
+static const int16_t kMinimumDifference[kNumChannels] = {
+    544, 544, 576, 576, 576, 576 };
+// Upper limit of mean value for speech model, Q7
+static const int16_t kMaximumSpeech[kNumChannels] = {
+    11392, 11392, 11520, 11520, 11520, 11520 };
+// Minimum value for mean value
+static const int16_t kMinimumMean[kNumGaussians] = { 640, 768 };
+// Upper limit of mean value for noise model, Q7
+static const int16_t kMaximumNoise[kNumChannels] = {
+    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 int16_t kNoiseDataWeights[kTableSize] = {
+    34, 62, 72, 66, 53, 25, 94, 66, 56, 62, 75, 103 };
+// Weights for the two Gaussians for the six channels (speech)
+static const int16_t kSpeechDataWeights[kTableSize] = {
+    48, 82, 45, 87, 50, 47, 80, 46, 83, 41, 78, 81 };
+// Means for the two Gaussians for the six channels (noise)
+static const int16_t kNoiseDataMeans[kTableSize] = {
+    6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863, 7820, 7266, 5020, 4362 };
+// Means for the two Gaussians for the six channels (speech)
+static const int16_t kSpeechDataMeans[kTableSize] = {
+    8306, 10085, 10078, 11823, 11843, 6309, 9473, 9571, 10879, 7581, 8180, 7483
+};
+// Stds for the two Gaussians for the six channels (noise)
+static const int16_t kNoiseDataStds[kTableSize] = {
+    378, 1064, 493, 582, 688, 593, 474, 697, 475, 688, 421, 455 };
+// Stds for the two Gaussians for the six channels (speech)
+static const int16_t kSpeechDataStds[kTableSize] = {
+    555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540, 1079, 850 };
+
+// Constants used in GmmProbability().
+//
+// Maximum number of counted speech (VAD = 1) frames in a row.
+static const int16_t kMaxSpeechFrames = 6;
+// Minimum standard deviation for both speech and noise.
+static const int16_t kMinStd = 384;
+
+// Constants in WebRtcVad_InitCore().
+// Default aggressiveness mode.
+static const short kDefaultMode = 0;
+static const int kInitCheck = 42;
+
+// Constants used in WebRtcVad_set_mode_core().
+//
+// Thresholds for different frame lengths (10 ms, 20 ms and 30 ms).
+//
+// Mode 0, Quality.
+static const int16_t kOverHangMax1Q[3] = { 8, 4, 3 };
+static const int16_t kOverHangMax2Q[3] = { 14, 7, 5 };
+static const int16_t kLocalThresholdQ[3] = { 24, 21, 24 };
+static const int16_t kGlobalThresholdQ[3] = { 57, 48, 57 };
+// Mode 1, Low bitrate.
+static const int16_t kOverHangMax1LBR[3] = { 8, 4, 3 };
+static const int16_t kOverHangMax2LBR[3] = { 14, 7, 5 };
+static const int16_t kLocalThresholdLBR[3] = { 37, 32, 37 };
+static const int16_t kGlobalThresholdLBR[3] = { 100, 80, 100 };
+// Mode 2, Aggressive.
+static const int16_t kOverHangMax1AGG[3] = { 6, 3, 2 };
+static const int16_t kOverHangMax2AGG[3] = { 9, 5, 3 };
+static const int16_t kLocalThresholdAGG[3] = { 82, 78, 82 };
+static const int16_t kGlobalThresholdAGG[3] = { 285, 260, 285 };
+// Mode 3, Very aggressive.
+static const int16_t kOverHangMax1VAG[3] = { 6, 3, 2 };
+static const int16_t kOverHangMax2VAG[3] = { 9, 5, 3 };
+static const int16_t kLocalThresholdVAG[3] = { 94, 94, 94 };
+static const int16_t kGlobalThresholdVAG[3] = { 1100, 1050, 1100 };
+
+// Calculates the weighted average w.r.t. number of Gaussians. The |data| are
+// updated with an |offset| before averaging.
+//
+// - data     [i/o] : Data to average.
+// - offset   [i]   : An offset added to |data|.
+// - weights  [i]   : Weights used for averaging.
+//
+// returns          : The weighted average.
+static int32_t WeightedAverage(int16_t* data, int16_t offset,
+                               const int16_t* weights) {
+  int k;
+  int32_t weighted_average = 0;
+
+  for (k = 0; k < kNumGaussians; k++) {
+    data[k * kNumChannels] += offset;
+    weighted_average += data[k * kNumChannels] * weights[k * kNumChannels];
+  }
+  return weighted_average;
+}
+
+// Calculates the probabilities for both speech and background noise using
+// Gaussian Mixture Models (GMM). A hypothesis-test is performed to decide which
+// type of signal is most probable.
+//
+// - self           [i/o] : Pointer to VAD instance
+// - features       [i]   : Feature vector of length |kNumChannels|
+//                          = log10(energy in frequency band)
+// - total_power    [i]   : Total power in audio frame.
+// - frame_length   [i]   : Number of input samples
+//
+// - returns              : the VAD decision (0 - noise, 1 - speech).
+static int16_t GmmProbability(VadInstT* self, int16_t* features,
+                              int16_t total_power, int frame_length) {
+  int channel, k;
+  int16_t feature_minimum;
+  int16_t h0, h1;
+  int16_t log_likelihood_ratio;
+  int16_t vadflag = 0;
+  int16_t shifts_h0, shifts_h1;
+  int16_t tmp_s16, tmp1_s16, tmp2_s16;
+  int16_t diff;
+  int gaussian;
+  int16_t nmk, nmk2, nmk3, smk, smk2, nsk, ssk;
+  int16_t delt, ndelt;
+  int16_t maxspe, maxmu;
+  int16_t deltaN[kTableSize], deltaS[kTableSize];
+  int16_t ngprvec[kTableSize] = { 0 };  // Conditional probability = 0.
+  int16_t sgprvec[kTableSize] = { 0 };  // Conditional probability = 0.
+  int32_t h0_test, h1_test;
+  int32_t tmp1_s32, tmp2_s32;
+  int32_t sum_log_likelihood_ratios = 0;
+  int32_t noise_global_mean, speech_global_mean;
+  int32_t noise_probability[kNumGaussians], speech_probability[kNumGaussians];
+  int16_t overhead1, overhead2, individualTest, totalTest;
+
+  // Set various thresholds based on frame lengths (80, 160 or 240 samples).
+  if (frame_length == 80) {
+    overhead1 = self->over_hang_max_1[0];
+    overhead2 = self->over_hang_max_2[0];
+    individualTest = self->individual[0];
+    totalTest = self->total[0];
+  } else if (frame_length == 160) {
+    overhead1 = self->over_hang_max_1[1];
+    overhead2 = self->over_hang_max_2[1];
+    individualTest = self->individual[1];
+    totalTest = self->total[1];
+  } else {
+    overhead1 = self->over_hang_max_1[2];
+    overhead2 = self->over_hang_max_2[2];
+    individualTest = self->individual[2];
+    totalTest = self->total[2];
+  }
+
+  if (total_power > kMinEnergy) {
+    // The signal power of current frame is large enough for processing. The
+    // processing consists of two parts:
+    // 1) Calculating the likelihood of speech and thereby a VAD decision.
+    // 2) Updating the underlying model, w.r.t., the decision made.
+
+    // The detection scheme is an LRT with hypothesis
+    // H0: Noise
+    // H1: Speech
+    //
+    // We combine a global LRT with local tests, for each frequency sub-band,
+    // here defined as |channel|.
+    for (channel = 0; channel < kNumChannels; channel++) {
+      // For each channel we model the probability with a GMM consisting of
+      // |kNumGaussians|, with different means and standard deviations depending
+      // on H0 or H1.
+      h0_test = 0;
+      h1_test = 0;
+      for (k = 0; k < kNumGaussians; k++) {
+        gaussian = channel + k * kNumChannels;
+        // Probability under H0, that is, probability of frame being noise.
+        // Value given in Q27 = Q7 * Q20.
+        tmp1_s32 = WebRtcVad_GaussianProbability(features[channel],
+                                                 self->noise_means[gaussian],
+                                                 self->noise_stds[gaussian],
+                                                 &deltaN[gaussian]);
+        noise_probability[k] = kNoiseDataWeights[gaussian] * tmp1_s32;
+        h0_test += noise_probability[k];  // Q27
+
+        // Probability under H1, that is, probability of frame being speech.
+        // Value given in Q27 = Q7 * Q20.
+        tmp1_s32 = WebRtcVad_GaussianProbability(features[channel],
+                                                 self->speech_means[gaussian],
+                                                 self->speech_stds[gaussian],
+                                                 &deltaS[gaussian]);
+        speech_probability[k] = kSpeechDataWeights[gaussian] * tmp1_s32;
+        h1_test += speech_probability[k];  // Q27
+      }
+
+      // Calculate the log likelihood ratio: log2(Pr{X|H1} / Pr{X|H1}).
+      // Approximation:
+      // log2(Pr{X|H1} / Pr{X|H1}) = log2(Pr{X|H1}*2^Q) - log2(Pr{X|H1}*2^Q)
+      //                           = log2(h1_test) - log2(h0_test)
+      //                           = log2(2^(31-shifts_h1)*(1+b1))
+      //                             - log2(2^(31-shifts_h0)*(1+b0))
+      //                           = shifts_h0 - shifts_h1
+      //                             + log2(1+b1) - log2(1+b0)
+      //                          ~= shifts_h0 - shifts_h1
+      //
+      // Note that b0 and b1 are values less than 1, hence, 0 <= log2(1+b0) < 1.
+      // Further, b0 and b1 are independent and on the average the two terms
+      // cancel.
+      shifts_h0 = WebRtcSpl_NormW32(h0_test);
+      shifts_h1 = WebRtcSpl_NormW32(h1_test);
+      if (h0_test == 0) {
+        shifts_h0 = 31;
+      }
+      if (h1_test == 0) {
+        shifts_h1 = 31;
+      }
+      log_likelihood_ratio = shifts_h0 - shifts_h1;
+
+      // Update |sum_log_likelihood_ratios| with spectrum weighting. This is
+      // used for the global VAD decision.
+      sum_log_likelihood_ratios +=
+          (int32_t) (log_likelihood_ratio * kSpectrumWeight[channel]);
+
+      // Local VAD decision.
+      if ((log_likelihood_ratio << 2) > individualTest) {
+        vadflag = 1;
+      }
+
+      // TODO(bjornv): The conditional probabilities below are applied on the
+      // hard coded number of Gaussians set to two. Find a way to generalize.
+      // Calculate local noise probabilities used later when updating the GMM.
+      h0 = (int16_t) (h0_test >> 12);  // Q15
+      if (h0 > 0) {
+        // High probability of noise. Assign conditional probabilities for each
+        // Gaussian in the GMM.
+        tmp1_s32 = (noise_probability[0] & 0xFFFFF000) << 2;  // Q29
+        ngprvec[channel] = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, h0);  // Q14
+        ngprvec[channel + kNumChannels] = 16384 - ngprvec[channel];
+      } else {
+        // Low noise probability. Assign conditional probability 1 to the first
+        // Gaussian and 0 to the rest (which is already set at initialization).
+        ngprvec[channel] = 16384;
+      }
+
+      // Calculate local speech probabilities used later when updating the GMM.
+      h1 = (int16_t) (h1_test >> 12);  // Q15
+      if (h1 > 0) {
+        // High probability of speech. Assign conditional probabilities for each
+        // Gaussian in the GMM. Otherwise use the initialized values, i.e., 0.
+        tmp1_s32 = (speech_probability[0] & 0xFFFFF000) << 2;  // Q29
+        sgprvec[channel] = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, h1);  // Q14
+        sgprvec[channel + kNumChannels] = 16384 - sgprvec[channel];
+      }
+    }
+
+    // Make a global VAD decision.
+    vadflag |= (sum_log_likelihood_ratios >= totalTest);
+
+    // Update the model parameters.
+    maxspe = 12800;
+    for (channel = 0; channel < kNumChannels; channel++) {
+
+      // Get minimum value in past which is used for long term correction in Q4.
+      feature_minimum = WebRtcVad_FindMinimum(self, features[channel], channel);
+
+      // Compute the "global" mean, that is the sum of the two means weighted.
+      noise_global_mean = WeightedAverage(&self->noise_means[channel], 0,
+                                          &kNoiseDataWeights[channel]);
+      tmp1_s16 = (int16_t) (noise_global_mean >> 6);  // Q8
+
+      for (k = 0; k < kNumGaussians; k++) {
+        gaussian = channel + k * kNumChannels;
+
+        nmk = self->noise_means[gaussian];
+        smk = self->speech_means[gaussian];
+        nsk = self->noise_stds[gaussian];
+        ssk = self->speech_stds[gaussian];
+
+        // Update noise mean vector if the frame consists of noise only.
+        nmk2 = nmk;
+        if (!vadflag) {
+          // deltaN = (x-mu)/sigma^2
+          // ngprvec[k] = |noise_probability[k]| /
+          //   (|noise_probability[0]| + |noise_probability[1]|)
+
+          // (Q14 * Q11 >> 11) = Q14.
+          delt = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(ngprvec[gaussian],
+                                                     deltaN[gaussian],
+                                                     11);
+          // Q7 + (Q14 * Q15 >> 22) = Q7.
+          nmk2 = nmk + (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(delt,
+                                                           kNoiseUpdateConst,
+                                                           22);
+        }
+
+        // Long term correction of the noise mean.
+        // Q8 - Q8 = Q8.
+        ndelt = (feature_minimum << 4) - tmp1_s16;
+        // Q7 + (Q8 * Q8) >> 9 = Q7.
+        nmk3 = nmk2 + (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(ndelt, kBackEta, 9);
+
+        // Control that the noise mean does not drift to much.
+        tmp_s16 = (int16_t) ((k + 5) << 7);
+        if (nmk3 < tmp_s16) {
+          nmk3 = tmp_s16;
+        }
+        tmp_s16 = (int16_t) ((72 + k - channel) << 7);
+        if (nmk3 > tmp_s16) {
+          nmk3 = tmp_s16;
+        }
+        self->noise_means[gaussian] = nmk3;
+
+        if (vadflag) {
+          // Update speech mean vector:
+          // |deltaS| = (x-mu)/sigma^2
+          // sgprvec[k] = |speech_probability[k]| /
+          //   (|speech_probability[0]| + |speech_probability[1]|)
+
+          // (Q14 * Q11) >> 11 = Q14.
+          delt = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(sgprvec[gaussian],
+                                                     deltaS[gaussian],
+                                                     11);
+          // Q14 * Q15 >> 21 = Q8.
+          tmp_s16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(delt,
+                                                        kSpeechUpdateConst,
+                                                        21);
+          // Q7 + (Q8 >> 1) = Q7. With rounding.
+          smk2 = smk + ((tmp_s16 + 1) >> 1);
+
+          // 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;
+          }
+          self->speech_means[gaussian] = smk2;  // Q7.
+
+          // (Q7 >> 3) = Q4. With rounding.
+          tmp_s16 = ((smk + 4) >> 3);
+
+          tmp_s16 = features[channel] - tmp_s16;  // Q4
+          // (Q11 * Q4 >> 3) = Q12.
+          tmp1_s32 = WEBRTC_SPL_MUL_16_16_RSFT(deltaS[gaussian], tmp_s16, 3);
+          tmp2_s32 = tmp1_s32 - 4096;
+          tmp_s16 = sgprvec[gaussian] >> 2;
+          // (Q14 >> 2) * Q12 = Q24.
+          tmp1_s32 = tmp_s16 * tmp2_s32;
+
+          tmp2_s32 = tmp1_s32 >> 4;  // Q20
+
+          // 0.1 * Q20 / Q7 = Q13.
+          if (tmp2_s32 > 0) {
+            tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(tmp2_s32, ssk * 10);
+          } else {
+            tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(-tmp2_s32, ssk * 10);
+            tmp_s16 = -tmp_s16;
+          }
+          // Divide by 4 giving an update factor of 0.025 (= 0.1 / 4).
+          // Note that division by 4 equals shift by 2, hence,
+          // (Q13 >> 8) = (Q13 >> 6) / 4 = Q7.
+          tmp_s16 += 128;  // Rounding.
+          ssk += (tmp_s16 >> 8);
+          if (ssk < kMinStd) {
+            ssk = kMinStd;
+          }
+          self->speech_stds[gaussian] = ssk;
+        } else {
+          // Update GMM variance vectors.
+          // deltaN * (features[channel] - nmk) - 1
+          // Q4 - (Q7 >> 3) = Q4.
+          tmp_s16 = features[channel] - (nmk >> 3);
+          // (Q11 * Q4 >> 3) = Q12.
+          tmp1_s32 = WEBRTC_SPL_MUL_16_16_RSFT(deltaN[gaussian], tmp_s16, 3);
+          tmp1_s32 -= 4096;
+
+          // (Q14 >> 2) * Q12 = Q24.
+          tmp_s16 = (ngprvec[gaussian] + 2) >> 2;
+          tmp2_s32 = tmp_s16 * tmp1_s32;
+          // Q20  * approx 0.001 (2^-10=0.0009766), hence,
+          // (Q24 >> 14) = (Q24 >> 4) / 2^10 = Q20.
+          tmp1_s32 = tmp2_s32 >> 14;
+
+          // Q20 / Q7 = Q13.
+          if (tmp1_s32 > 0) {
+            tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(tmp1_s32, nsk);
+          } else {
+            tmp_s16 = (int16_t) WebRtcSpl_DivW32W16(-tmp1_s32, nsk);
+            tmp_s16 = -tmp_s16;
+          }
+          tmp_s16 += 32;  // Rounding
+          nsk += tmp_s16 >> 6;  // Q13 >> 6 = Q7.
+          if (nsk < kMinStd) {
+            nsk = kMinStd;
+          }
+          self->noise_stds[gaussian] = nsk;
+        }
+      }
+
+      // Separate models if they are too close.
+      // |noise_global_mean| in Q14 (= Q7 * Q7).
+      noise_global_mean = WeightedAverage(&self->noise_means[channel], 0,
+                                          &kNoiseDataWeights[channel]);
+
+      // |speech_global_mean| in Q14 (= Q7 * Q7).
+      speech_global_mean = WeightedAverage(&self->speech_means[channel], 0,
+                                           &kSpeechDataWeights[channel]);
+
+      // |diff| = "global" speech mean - "global" noise mean.
+      // (Q14 >> 9) - (Q14 >> 9) = Q5.
+      diff = (int16_t) (speech_global_mean >> 9) -
+          (int16_t) (noise_global_mean >> 9);
+      if (diff < kMinimumDifference[channel]) {
+        tmp_s16 = kMinimumDifference[channel] - diff;
+
+        // |tmp1_s16| = ~0.8 * (kMinimumDifference - diff) in Q7.
+        // |tmp2_s16| = ~0.2 * (kMinimumDifference - diff) in Q7.
+        tmp1_s16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(13, tmp_s16, 2);
+        tmp2_s16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(3, tmp_s16, 2);
+
+        // Move Gaussian means for speech model by |tmp1_s16| and update
+        // |speech_global_mean|. Note that |self->speech_means[channel]| is
+        // changed after the call.
+        speech_global_mean = WeightedAverage(&self->speech_means[channel],
+                                             tmp1_s16,
+                                             &kSpeechDataWeights[channel]);
+
+        // Move Gaussian means for noise model by -|tmp2_s16| and update
+        // |noise_global_mean|. Note that |self->noise_means[channel]| is
+        // changed after the call.
+        noise_global_mean = WeightedAverage(&self->noise_means[channel],
+                                            -tmp2_s16,
+                                            &kNoiseDataWeights[channel]);
+      }
+
+      // Control that the speech & noise means do not drift to much.
+      maxspe = kMaximumSpeech[channel];
+      tmp2_s16 = (int16_t) (speech_global_mean >> 7);
+      if (tmp2_s16 > maxspe) {
+        // Upper limit of speech model.
+        tmp2_s16 -= maxspe;
+
+        for (k = 0; k < kNumGaussians; k++) {
+          self->speech_means[channel + k * kNumChannels] -= tmp2_s16;
+        }
+      }
+
+      tmp2_s16 = (int16_t) (noise_global_mean >> 7);
+      if (tmp2_s16 > kMaximumNoise[channel]) {
+        tmp2_s16 -= kMaximumNoise[channel];
+
+        for (k = 0; k < kNumGaussians; k++) {
+          self->noise_means[channel + k * kNumChannels] -= tmp2_s16;
+        }
+      }
+    }
+    self->frame_counter++;
+  }
+
+  // Smooth with respect to transition hysteresis.
+  if (!vadflag) {
+    if (self->over_hang > 0) {
+      vadflag = 2 + self->over_hang;
+      self->over_hang--;
+    }
+    self->num_of_speech = 0;
+  } else {
+    self->num_of_speech++;
+    if (self->num_of_speech > kMaxSpeechFrames) {
+      self->num_of_speech = kMaxSpeechFrames;
+      self->over_hang = overhead2;
+    } else {
+      self->over_hang = overhead1;
+    }
+  }
+  return vadflag;
+}
+
+// Initialize the VAD. Set aggressiveness mode to default value.
+int WebRtcVad_InitCore(VadInstT* self) {
+  int i;
+
+  if (self == NULL) {
+    return -1;
+  }
+
+  // Initialization of general struct variables.
+  self->vad = 1;  // Speech active (=1).
+  self->frame_counter = 0;
+  self->over_hang = 0;
+  self->num_of_speech = 0;
+
+  // Initialization of downsampling filter state.
+  memset(self->downsampling_filter_states, 0,
+         sizeof(self->downsampling_filter_states));
+
+  // Initialization of 48 to 8 kHz downsampling.
+  WebRtcSpl_ResetResample48khzTo8khz(&self->state_48_to_8);
+
+  // Read initial PDF parameters.
+  for (i = 0; i < kTableSize; i++) {
+    self->noise_means[i] = kNoiseDataMeans[i];
+    self->speech_means[i] = kSpeechDataMeans[i];
+    self->noise_stds[i] = kNoiseDataStds[i];
+    self->speech_stds[i] = kSpeechDataStds[i];
+  }
+
+  // Initialize Index and Minimum value vectors.
+  for (i = 0; i < 16 * kNumChannels; i++) {
+    self->low_value_vector[i] = 10000;
+    self->index_vector[i] = 0;
+  }
+
+  // Initialize splitting filter states.
+  memset(self->upper_state, 0, sizeof(self->upper_state));
+  memset(self->lower_state, 0, sizeof(self->lower_state));
+
+  // Initialize high pass filter states.
+  memset(self->hp_filter_state, 0, sizeof(self->hp_filter_state));
+
+  // Initialize mean value memory, for WebRtcVad_FindMinimum().
+  for (i = 0; i < kNumChannels; i++) {
+    self->mean_value[i] = 1600;
+  }
+
+  // Set aggressiveness mode to default (=|kDefaultMode|).
+  if (WebRtcVad_set_mode_core(self, kDefaultMode) != 0) {
+    return -1;
+  }
+
+  self->init_flag = kInitCheck;
+
+  return 0;
+}
+
+// Set aggressiveness mode
+int WebRtcVad_set_mode_core(VadInstT* self, int mode) {
+  int return_value = 0;
+
+  switch (mode) {
+    case 0:
+      // Quality mode.
+      memcpy(self->over_hang_max_1, kOverHangMax1Q,
+             sizeof(self->over_hang_max_1));
+      memcpy(self->over_hang_max_2, kOverHangMax2Q,
+             sizeof(self->over_hang_max_2));
+      memcpy(self->individual, kLocalThresholdQ,
+             sizeof(self->individual));
+      memcpy(self->total, kGlobalThresholdQ,
+             sizeof(self->total));
+      break;
+    case 1:
+      // Low bitrate mode.
+      memcpy(self->over_hang_max_1, kOverHangMax1LBR,
+             sizeof(self->over_hang_max_1));
+      memcpy(self->over_hang_max_2, kOverHangMax2LBR,
+             sizeof(self->over_hang_max_2));
+      memcpy(self->individual, kLocalThresholdLBR,
+             sizeof(self->individual));
+      memcpy(self->total, kGlobalThresholdLBR,
+             sizeof(self->total));
+      break;
+    case 2:
+      // Aggressive mode.
+      memcpy(self->over_hang_max_1, kOverHangMax1AGG,
+             sizeof(self->over_hang_max_1));
+      memcpy(self->over_hang_max_2, kOverHangMax2AGG,
+             sizeof(self->over_hang_max_2));
+      memcpy(self->individual, kLocalThresholdAGG,
+             sizeof(self->individual));
+      memcpy(self->total, kGlobalThresholdAGG,
+             sizeof(self->total));
+      break;
+    case 3:
+      // Very aggressive mode.
+      memcpy(self->over_hang_max_1, kOverHangMax1VAG,
+             sizeof(self->over_hang_max_1));
+      memcpy(self->over_hang_max_2, kOverHangMax2VAG,
+             sizeof(self->over_hang_max_2));
+      memcpy(self->individual, kLocalThresholdVAG,
+             sizeof(self->individual));
+      memcpy(self->total, kGlobalThresholdVAG,
+             sizeof(self->total));
+      break;
+    default:
+      return_value = -1;
+      break;
+  }
+
+  return return_value;
+}
+
+// Calculate VAD decision by first extracting feature values and then calculate
+// probability for both speech and background noise.
+
+int WebRtcVad_CalcVad48khz(VadInstT* inst, int16_t* speech_frame,
+                           int frame_length) {
+  int vad;
+  int i;
+  int16_t speech_nb[240];  // 30 ms in 8 kHz.
+  // |tmp_mem| is a temporary memory used by resample function, length is
+  // frame length in 10 ms (480 samples) + 256 extra.
+  int32_t tmp_mem[480 + 256] = { 0 };
+  const int kFrameLen10ms48khz = 480;
+  const int kFrameLen10ms8khz = 80;
+  int num_10ms_frames = frame_length / kFrameLen10ms48khz;
+
+  for (i = 0; i < num_10ms_frames; i++) {
+    WebRtcSpl_Resample48khzTo8khz(speech_frame,
+                                  &speech_nb[i * kFrameLen10ms8khz],
+                                  &inst->state_48_to_8,
+                                  tmp_mem);
+  }
+
+  // Do VAD on an 8 kHz signal
+  vad = WebRtcVad_CalcVad8khz(inst, speech_nb, frame_length / 6);
+
+  return vad;
+}
+
+int WebRtcVad_CalcVad32khz(VadInstT* inst, int16_t* speech_frame,
+                           int frame_length)
+{
+    int len, vad;
+    int16_t speechWB[480]; // Downsampled speech frame: 960 samples (30ms in SWB)
+    int16_t 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;
+}
+
+int WebRtcVad_CalcVad16khz(VadInstT* inst, int16_t* speech_frame,
+                           int frame_length)
+{
+    int len, vad;
+    int16_t 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;
+}
+
+int WebRtcVad_CalcVad8khz(VadInstT* inst, int16_t* speech_frame,
+                          int frame_length)
+{
+    int16_t feature_vector[kNumChannels], total_power;
+
+    // Get power in the bands
+    total_power = WebRtcVad_CalculateFeatures(inst, speech_frame, frame_length,
+                                              feature_vector);
+
+    // Make a VAD
+    inst->vad = GmmProbability(inst, feature_vector, total_power, frame_length);
+
+    return inst->vad;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_core.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_core.h
new file mode 100644
index 0000000..0a687c7
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_core.h
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2012 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_COMMON_AUDIO_VAD_VAD_CORE_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
+
+#include "../signal_processing/include/signal_processing_library.h"
+#include "../include/typedefs.h"
+
+enum { kNumChannels = 6 };  // Number of frequency bands (named channels).
+enum { kNumGaussians = 2 };  // Number of Gaussians per channel in the GMM.
+enum { kTableSize = kNumChannels * kNumGaussians };
+enum { kMinEnergy = 10 };  // Minimum energy required to trigger audio signal.
+
+typedef struct VadInstT_
+{
+
+    int vad;
+    int32_t downsampling_filter_states[4];
+    WebRtcSpl_State48khzTo8khz state_48_to_8;
+    int16_t noise_means[kTableSize];
+    int16_t speech_means[kTableSize];
+    int16_t noise_stds[kTableSize];
+    int16_t speech_stds[kTableSize];
+    // TODO(bjornv): Change to |frame_count|.
+    int32_t frame_counter;
+    int16_t over_hang; // Over Hang
+    int16_t num_of_speech;
+    // TODO(bjornv): Change to |age_vector|.
+    int16_t index_vector[16 * kNumChannels];
+    int16_t low_value_vector[16 * kNumChannels];
+    // TODO(bjornv): Change to |median|.
+    int16_t mean_value[kNumChannels];
+    int16_t upper_state[5];
+    int16_t lower_state[5];
+    int16_t hp_filter_state[4];
+    int16_t over_hang_max_1[3];
+    int16_t over_hang_max_2[3];
+    int16_t individual[3];
+    int16_t total[3];
+
+    int init_flag;
+
+} VadInstT;
+
+// Initializes the core VAD component. The default aggressiveness mode is
+// controlled by |kDefaultMode| in vad_core.c.
+//
+// - self [i/o] : Instance that should be initialized
+//
+// returns      : 0 (OK), -1 (NULL pointer in or if the default mode can't be
+//                set)
+int WebRtcVad_InitCore(VadInstT* self);
+
+/****************************************************************************
+ * 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* self, int mode);
+
+/****************************************************************************
+ * WebRtcVad_CalcVad48khz(...)
+ * 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
+ */
+int WebRtcVad_CalcVad48khz(VadInstT* inst, int16_t* speech_frame,
+                           int frame_length);
+int WebRtcVad_CalcVad32khz(VadInstT* inst, int16_t* speech_frame,
+                           int frame_length);
+int WebRtcVad_CalcVad16khz(VadInstT* inst, int16_t* speech_frame,
+                           int frame_length);
+int WebRtcVad_CalcVad8khz(VadInstT* inst, int16_t* speech_frame,
+                          int frame_length);
+
+#endif  // WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_filterbank.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_filterbank.c
new file mode 100644
index 0000000..1796667
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_filterbank.c
@@ -0,0 +1,334 @@
+/*
+ *  Copyright (c) 2012 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 "vad_filterbank.h"
+
+#include <assert.h>
+
+#include "../signal_processing/include/signal_processing_library.h"
+#include "../include/typedefs.h"
+
+// Constants used in LogOfEnergy().
+static const int16_t kLogConst = 24660;  // 160*log10(2) in Q9.
+static const int16_t kLogEnergyIntPart = 14336;  // 14 in Q10
+
+// Coefficients used by HighPassFilter, Q14.
+static const int16_t kHpZeroCoefs[3] = { 6631, -13262, 6631 };
+static const int16_t kHpPoleCoefs[3] = { 16384, -7756, 5620 };
+
+// Allpass filter coefficients, upper and lower, in Q15.
+// Upper: 0.64, Lower: 0.17
+static const int16_t kAllPassCoefsQ15[2] = { 20972, 5571 };
+
+// Adjustment for division with two in SplitFilter.
+static const int16_t kOffsetVector[6] = { 368, 368, 272, 176, 176, 176 };
+
+// High pass filtering, with a cut-off frequency at 80 Hz, if the |data_in| is
+// sampled at 500 Hz.
+//
+// - data_in      [i]   : Input audio data sampled at 500 Hz.
+// - data_length  [i]   : Length of input and output data.
+// - filter_state [i/o] : State of the filter.
+// - data_out     [o]   : Output audio data in the frequency interval
+//                        80 - 250 Hz.
+static void HighPassFilter(const int16_t* data_in, int data_length,
+                           int16_t* filter_state, int16_t* data_out) {
+  int i;
+  const int16_t* in_ptr = data_in;
+  int16_t* out_ptr = data_out;
+  int32_t tmp32 = 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 < data_length; i++) {
+    // All-zero section (filter coefficients in Q14).
+    tmp32 = WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[0], *in_ptr);
+    tmp32 += WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[1], filter_state[0]);
+    tmp32 += WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[2], filter_state[1]);
+    filter_state[1] = filter_state[0];
+    filter_state[0] = *in_ptr++;
+
+    // All-pole section (filter coefficients in Q14).
+    tmp32 -= WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[1], filter_state[2]);
+    tmp32 -= WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[2], filter_state[3]);
+    filter_state[3] = filter_state[2];
+    filter_state[2] = (int16_t) (tmp32 >> 14);
+    *out_ptr++ = filter_state[2];
+  }
+}
+
+// All pass filtering of |data_in|, used before splitting the signal into two
+// frequency bands (low pass vs high pass).
+// Note that |data_in| and |data_out| can NOT correspond to the same address.
+//
+// - data_in            [i]   : Input audio signal given in Q0.
+// - data_length        [i]   : Length of input and output data.
+// - filter_coefficient [i]   : Given in Q15.
+// - filter_state       [i/o] : State of the filter given in Q(-1).
+// - data_out           [o]   : Output audio signal given in Q(-1).
+static void AllPassFilter(const int16_t* data_in, int data_length,
+                          int16_t filter_coefficient, int16_t* filter_state,
+                          int16_t* data_out) {
+  // 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 i;
+  int16_t tmp16 = 0;
+  int32_t tmp32 = 0;
+  int32_t state32 = ((int32_t) (*filter_state) << 16);  // Q15
+
+  for (i = 0; i < data_length; i++) {
+    tmp32 = state32 + WEBRTC_SPL_MUL_16_16(filter_coefficient, *data_in);
+    tmp16 = (int16_t) (tmp32 >> 16);  // Q(-1)
+    *data_out++ = tmp16;
+    state32 = (((int32_t) (*data_in)) << 14); // Q14
+    state32 -= WEBRTC_SPL_MUL_16_16(filter_coefficient, tmp16);  // Q14
+    state32 <<= 1;  // Q15.
+    data_in += 2;
+  }
+
+  *filter_state = (int16_t) (state32 >> 16);  // Q(-1)
+}
+
+// Splits |data_in| into |hp_data_out| and |lp_data_out| corresponding to
+// an upper (high pass) part and a lower (low pass) part respectively.
+//
+// - data_in      [i]   : Input audio data to be split into two frequency bands.
+// - data_length  [i]   : Length of |data_in|.
+// - upper_state  [i/o] : State of the upper filter, given in Q(-1).
+// - lower_state  [i/o] : State of the lower filter, given in Q(-1).
+// - hp_data_out  [o]   : Output audio data of the upper half of the spectrum.
+//                        The length is |data_length| / 2.
+// - lp_data_out  [o]   : Output audio data of the lower half of the spectrum.
+//                        The length is |data_length| / 2.
+static void SplitFilter(const int16_t* data_in, int data_length,
+                        int16_t* upper_state, int16_t* lower_state,
+                        int16_t* hp_data_out, int16_t* lp_data_out) {
+  int i;
+  int half_length = data_length >> 1;  // Downsampling by 2.
+  int16_t tmp_out;
+
+  // All-pass filtering upper branch.
+  AllPassFilter(&data_in[0], half_length, kAllPassCoefsQ15[0], upper_state,
+                hp_data_out);
+
+  // All-pass filtering lower branch.
+  AllPassFilter(&data_in[1], half_length, kAllPassCoefsQ15[1], lower_state,
+                lp_data_out);
+
+  // Make LP and HP signals.
+  for (i = 0; i < half_length; i++) {
+    tmp_out = *hp_data_out;
+    *hp_data_out++ -= *lp_data_out;
+    *lp_data_out++ += tmp_out;
+  }
+}
+
+// Calculates the energy of |data_in| in dB, and also updates an overall
+// |total_energy| if necessary.
+//
+// - data_in      [i]   : Input audio data for energy calculation.
+// - data_length  [i]   : Length of input data.
+// - offset       [i]   : Offset value added to |log_energy|.
+// - total_energy [i/o] : An external energy updated with the energy of
+//                        |data_in|.
+//                        NOTE: |total_energy| is only updated if
+//                        |total_energy| <= |kMinEnergy|.
+// - log_energy   [o]   : 10 * log10("energy of |data_in|") given in Q4.
+static void LogOfEnergy(const int16_t* data_in, int data_length,
+                        int16_t offset, int16_t* total_energy,
+                        int16_t* log_energy) {
+  // |tot_rshifts| accumulates the number of right shifts performed on |energy|.
+  int tot_rshifts = 0;
+  // The |energy| will be normalized to 15 bits. We use unsigned integer because
+  // we eventually will mask out the fractional part.
+  uint32_t energy = 0;
+
+  assert(data_in != NULL);
+  assert(data_length > 0);
+
+  energy = (uint32_t) WebRtcSpl_Energy((int16_t*) data_in, data_length,
+                                       &tot_rshifts);
+
+  if (energy != 0) {
+    // By construction, normalizing to 15 bits is equivalent with 17 leading
+    // zeros of an unsigned 32 bit value.
+    int normalizing_rshifts = 17 - WebRtcSpl_NormU32(energy);
+    // In a 15 bit representation the leading bit is 2^14. log2(2^14) in Q10 is
+    // (14 << 10), which is what we initialize |log2_energy| with. For a more
+    // detailed derivations, see below.
+    int16_t log2_energy = kLogEnergyIntPart;
+
+    tot_rshifts += normalizing_rshifts;
+    // Normalize |energy| to 15 bits.
+    // |tot_rshifts| is now the total number of right shifts performed on
+    // |energy| after normalization. This means that |energy| is in
+    // Q(-tot_rshifts).
+    if (normalizing_rshifts < 0) {
+      energy <<= -normalizing_rshifts;
+    } else {
+      energy >>= normalizing_rshifts;
+    }
+
+    // Calculate the energy of |data_in| in dB, in Q4.
+    //
+    // 10 * log10("true energy") in Q4 = 2^4 * 10 * log10("true energy") =
+    // 160 * log10(|energy| * 2^|tot_rshifts|) =
+    // 160 * log10(2) * log2(|energy| * 2^|tot_rshifts|) =
+    // 160 * log10(2) * (log2(|energy|) + log2(2^|tot_rshifts|)) =
+    // (160 * log10(2)) * (log2(|energy|) + |tot_rshifts|) =
+    // |kLogConst| * (|log2_energy| + |tot_rshifts|)
+    //
+    // We know by construction that |energy| is normalized to 15 bits. Hence,
+    // |energy| = 2^14 + frac_Q15, where frac_Q15 is a fractional part in Q15.
+    // Further, we'd like |log2_energy| in Q10
+    // log2(|energy|) in Q10 = 2^10 * log2(2^14 + frac_Q15) =
+    // 2^10 * log2(2^14 * (1 + frac_Q15 * 2^-14)) =
+    // 2^10 * (14 + log2(1 + frac_Q15 * 2^-14)) ~=
+    // (14 << 10) + 2^10 * (frac_Q15 * 2^-14) =
+    // (14 << 10) + (frac_Q15 * 2^-4) = (14 << 10) + (frac_Q15 >> 4)
+    //
+    // Note that frac_Q15 = (|energy| & 0x00003FFF)
+
+    // Calculate and add the fractional part to |log2_energy|.
+    log2_energy += (int16_t) ((energy & 0x00003FFF) >> 4);
+
+    // |kLogConst| is in Q9, |log2_energy| in Q10 and |tot_rshifts| in Q0.
+    // Note that we in our derivation above have accounted for an output in Q4.
+    *log_energy = (int16_t) (WEBRTC_SPL_MUL_16_16_RSFT(
+        kLogConst, log2_energy, 19) +
+        WEBRTC_SPL_MUL_16_16_RSFT(tot_rshifts, kLogConst, 9));
+
+    if (*log_energy < 0) {
+      *log_energy = 0;
+    }
+  } else {
+    *log_energy = offset;
+    return;
+  }
+
+  *log_energy += offset;
+
+  // Update the approximate |total_energy| with the energy of |data_in|, if
+  // |total_energy| has not exceeded |kMinEnergy|. |total_energy| is used as an
+  // energy indicator in WebRtcVad_GmmProbability() in vad_core.c.
+  if (*total_energy <= kMinEnergy) {
+    if (tot_rshifts >= 0) {
+      // We know by construction that the |energy| > |kMinEnergy| in Q0, so add
+      // an arbitrary value such that |total_energy| exceeds |kMinEnergy|.
+      *total_energy += kMinEnergy + 1;
+    } else {
+      // By construction |energy| is represented by 15 bits, hence any number of
+      // right shifted |energy| will fit in an int16_t. In addition, adding the
+      // value to |total_energy| is wrap around safe as long as
+      // |kMinEnergy| < 8192.
+      *total_energy += (int16_t) (energy >> -tot_rshifts);  // Q0.
+    }
+  }
+}
+
+int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
+                                    int data_length, int16_t* features) {
+  int16_t total_energy = 0;
+  // We expect |data_length| to be 80, 160 or 240 samples, which corresponds to
+  // 10, 20 or 30 ms in 8 kHz. Therefore, the intermediate downsampled data will
+  // have at most 120 samples after the first split and at most 60 samples after
+  // the second split.
+  int16_t hp_120[120], lp_120[120];
+  int16_t hp_60[60], lp_60[60];
+  const int half_data_length = data_length >> 1;
+  int length = half_data_length;  // |data_length| / 2, corresponds to
+                                  // bandwidth = 2000 Hz after downsampling.
+
+  // Initialize variables for the first SplitFilter().
+  int frequency_band = 0;
+  const int16_t* in_ptr = data_in;  // [0 - 4000] Hz.
+  int16_t* hp_out_ptr = hp_120;  // [2000 - 4000] Hz.
+  int16_t* lp_out_ptr = lp_120;  // [0 - 2000] Hz.
+
+  assert(data_length >= 0);
+  assert(data_length <= 240);
+  assert(4 < kNumChannels - 1);  // Checking maximum |frequency_band|.
+
+  // Split at 2000 Hz and downsample.
+  SplitFilter(in_ptr, data_length, &self->upper_state[frequency_band],
+              &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+  // For the upper band (2000 Hz - 4000 Hz) split at 3000 Hz and downsample.
+  frequency_band = 1;
+  in_ptr = hp_120;  // [2000 - 4000] Hz.
+  hp_out_ptr = hp_60;  // [3000 - 4000] Hz.
+  lp_out_ptr = lp_60;  // [2000 - 3000] Hz.
+  SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+              &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+  // Energy in 3000 Hz - 4000 Hz.
+  length >>= 1;  // |data_length| / 4 <=> bandwidth = 1000 Hz.
+
+  LogOfEnergy(hp_60, length, kOffsetVector[5], &total_energy, &features[5]);
+
+  // Energy in 2000 Hz - 3000 Hz.
+  LogOfEnergy(lp_60, length, kOffsetVector[4], &total_energy, &features[4]);
+
+  // For the lower band (0 Hz - 2000 Hz) split at 1000 Hz and downsample.
+  frequency_band = 2;
+  in_ptr = lp_120;  // [0 - 2000] Hz.
+  hp_out_ptr = hp_60;  // [1000 - 2000] Hz.
+  lp_out_ptr = lp_60;  // [0 - 1000] Hz.
+  length = half_data_length;  // |data_length| / 2 <=> bandwidth = 2000 Hz.
+  SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+              &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+  // Energy in 1000 Hz - 2000 Hz.
+  length >>= 1;  // |data_length| / 4 <=> bandwidth = 1000 Hz.
+  LogOfEnergy(hp_60, length, kOffsetVector[3], &total_energy, &features[3]);
+
+  // For the lower band (0 Hz - 1000 Hz) split at 500 Hz and downsample.
+  frequency_band = 3;
+  in_ptr = lp_60;  // [0 - 1000] Hz.
+  hp_out_ptr = hp_120;  // [500 - 1000] Hz.
+  lp_out_ptr = lp_120;  // [0 - 500] Hz.
+  SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+              &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+  // Energy in 500 Hz - 1000 Hz.
+  length >>= 1;  // |data_length| / 8 <=> bandwidth = 500 Hz.
+  LogOfEnergy(hp_120, length, kOffsetVector[2], &total_energy, &features[2]);
+
+  // For the lower band (0 Hz - 500 Hz) split at 250 Hz and downsample.
+  frequency_band = 4;
+  in_ptr = lp_120;  // [0 - 500] Hz.
+  hp_out_ptr = hp_60;  // [250 - 500] Hz.
+  lp_out_ptr = lp_60;  // [0 - 250] Hz.
+  SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
+              &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
+
+  // Energy in 250 Hz - 500 Hz.
+  length >>= 1;  // |data_length| / 16 <=> bandwidth = 250 Hz.
+  LogOfEnergy(hp_60, length, kOffsetVector[1], &total_energy, &features[1]);
+
+  // Remove 0 Hz - 80 Hz, by high pass filtering the lower band.
+  HighPassFilter(lp_60, length, self->hp_filter_state, hp_120);
+
+  // Energy in 80 Hz - 250 Hz.
+  LogOfEnergy(hp_120, length, kOffsetVector[0], &total_energy, &features[0]);
+
+  return total_energy;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_filterbank.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_filterbank.h
new file mode 100644
index 0000000..27a916e
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_filterbank.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2012 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 feature calculating functionality used in vad_core.c.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
+
+#include "vad_core.h"
+#include "../include/typedefs.h"
+
+// Takes |data_length| samples of |data_in| and calculates the logarithm of the
+// energy of each of the |kNumChannels| = 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
+//
+// The values are given in Q4 and written to |features|. Further, an approximate
+// overall energy is returned. The return value is used in
+// WebRtcVad_GmmProbability() as a signal indicator, hence it is arbitrary above
+// the threshold |kMinEnergy|.
+//
+// - self         [i/o] : State information of the VAD.
+// - data_in      [i]   : Input audio data, for feature extraction.
+// - data_length  [i]   : Audio data size, in number of samples.
+// - features     [o]   : 10 * log10(energy in each frequency band), Q4.
+// - returns            : Total energy of the signal (NOTE! This value is not
+//                        exact. It is only used in a comparison.)
+int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
+                                    int data_length, int16_t* features);
+
+#endif  // WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_gmm.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_gmm.c
new file mode 100644
index 0000000..6d258bf
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_gmm.c
@@ -0,0 +1,83 @@
+/*
+ *  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 "vad_gmm.h"
+
+#include "../signal_processing/include/signal_processing_library.h"
+#include "../include/typedefs.h"
+
+static const int32_t kCompVar = 22005;
+static const int16_t kLog2Exp = 5909;  // log2(exp(1)) in Q12.
+
+// For a normal distribution, the probability of |input| is calculated and
+// returned (in Q20). The formula for normal distributed probability is
+//
+// 1 / s * exp(-(x - m)^2 / (2 * s^2))
+//
+// where the parameters are given in the following Q domains:
+// m = |mean| (Q7)
+// s = |std| (Q7)
+// x = |input| (Q4)
+// in addition to the probability we output |delta| (in Q11) used when updating
+// the noise/speech model.
+int32_t WebRtcVad_GaussianProbability(int16_t input,
+                                      int16_t mean,
+                                      int16_t std,
+                                      int16_t* delta) {
+  int16_t tmp16, inv_std, inv_std2, exp_value = 0;
+  int32_t tmp32;
+
+  // Calculate |inv_std| = 1 / s, in Q10.
+  // 131072 = 1 in Q17, and (|std| >> 1) is for rounding instead of truncation.
+  // Q-domain: Q17 / Q7 = Q10.
+  tmp32 = (int32_t) 131072 + (int32_t) (std >> 1);
+  inv_std = (int16_t) WebRtcSpl_DivW32W16(tmp32, std);
+
+  // Calculate |inv_std2| = 1 / s^2, in Q14.
+  tmp16 = (inv_std >> 2);  // Q10 -> Q8.
+  // Q-domain: (Q8 * Q8) >> 2 = Q14.
+  inv_std2 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 2);
+  // TODO(bjornv): Investigate if changing to
+  // |inv_std2| = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(|inv_std|, |inv_std|, 6);
+  // gives better accuracy.
+
+  tmp16 = (input << 3);  // Q4 -> Q7
+  tmp16 = tmp16 - mean;  // Q7 - Q7 = Q7
+
+  // To be used later, when updating noise/speech model.
+  // |delta| = (x - m) / s^2, in Q11.
+  // Q-domain: (Q14 * Q7) >> 10 = Q11.
+  *delta = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(inv_std2, tmp16, 10);
+
+  // Calculate the exponent |tmp32| = (x - m)^2 / (2 * s^2), in Q10. Replacing
+  // division by two with one shift.
+  // Q-domain: (Q11 * Q7) >> 8 = Q10.
+  tmp32 = WEBRTC_SPL_MUL_16_16_RSFT(*delta, tmp16, 9);
+
+  // If the exponent is small enough to give a non-zero probability we calculate
+  // |exp_value| ~= exp(-(x - m)^2 / (2 * s^2))
+  //             ~= exp2(-log2(exp(1)) * |tmp32|).
+  if (tmp32 < kCompVar) {
+    // Calculate |tmp16| = log2(exp(1)) * |tmp32|, in Q10.
+    // Q-domain: (Q12 * Q10) >> 12 = Q10.
+    tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(kLog2Exp, (int16_t) tmp32, 12);
+    tmp16 = -tmp16;
+    exp_value = (0x0400 | (tmp16 & 0x03FF));
+    tmp16 ^= 0xFFFF;
+    tmp16 >>= 10;
+    tmp16 += 1;
+    // Get |exp_value| = exp(-|tmp32|) in Q10.
+    exp_value >>= tmp16;
+  }
+
+  // Calculate and return (1 / s) * exp(-(x - m)^2 / (2 * s^2)), in Q20.
+  // Q-domain: Q10 * Q10 = Q20.
+  return WEBRTC_SPL_MUL_16_16(inv_std, exp_value);
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_gmm.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_gmm.h
new file mode 100644
index 0000000..62390b2
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_gmm.h
@@ -0,0 +1,39 @@
+/*
+ *  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.
+ */
+
+// Gaussian probability calculations internally used in vad_core.c.
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
+
+#include "../include/typedefs.h"
+
+// Calculates the probability for |input|, given that |input| comes from a
+// normal distribution with mean and standard deviation (|mean|, |std|).
+//
+// Inputs:
+//      - input         : input sample in Q4.
+//      - mean          : mean input in the statistical model, Q7.
+//      - std           : standard deviation, Q7.
+//
+// Output:
+//
+//      - delta         : input used when updating the model, Q11.
+//                        |delta| = (|input| - |mean|) / |std|^2.
+//
+// Return:
+//   (probability for |input|) =
+//    1 / |std| * exp(-(|input| - |mean|)^2 / (2 * |std|^2));
+int32_t WebRtcVad_GaussianProbability(int16_t input,
+                                      int16_t mean,
+                                      int16_t std,
+                                      int16_t* delta);
+
+#endif  // WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_sp.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_sp.c
new file mode 100644
index 0000000..d0c1f1c
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_sp.c
@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2012 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 "vad_sp.h"
+
+#include <assert.h>
+
+#include "../signal_processing/include/signal_processing_library.h"
+#include "vad_core.h"
+#include "../include/typedefs.h"
+
+// Allpass filter coefficients, upper and lower, in Q13.
+// Upper: 0.64, Lower: 0.17.
+static const int16_t kAllPassCoefsQ13[2] = { 5243, 1392 };  // Q13.
+static const int16_t kSmoothingDown = 6553;  // 0.2 in Q15.
+static const int16_t kSmoothingUp = 32439;  // 0.99 in Q15.
+
+// TODO(bjornv): Move this function to vad_filterbank.c.
+// Downsampling filter based on splitting filter and allpass functions.
+void WebRtcVad_Downsampling(int16_t* signal_in,
+                            int16_t* signal_out,
+                            int32_t* filter_state,
+                            int in_length) {
+  int16_t tmp16_1 = 0, tmp16_2 = 0;
+  int32_t tmp32_1 = filter_state[0];
+  int32_t tmp32_2 = filter_state[1];
+  int n = 0;
+  int half_length = (in_length >> 1);  // Downsampling by 2 gives half length.
+
+  // Filter coefficients in Q13, filter state in Q0.
+  for (n = 0; n < half_length; n++) {
+    // All-pass filtering upper branch.
+    tmp16_1 = (int16_t) ((tmp32_1 >> 1) +
+        WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], *signal_in, 14));
+    *signal_out = tmp16_1;
+    tmp32_1 = (int32_t) (*signal_in++) -
+        WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], tmp16_1, 12);
+
+    // All-pass filtering lower branch.
+    tmp16_2 = (int16_t) ((tmp32_2 >> 1) +
+        WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], *signal_in, 14));
+    *signal_out++ += tmp16_2;
+    tmp32_2 = (int32_t) (*signal_in++) -
+        WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], tmp16_2, 12);
+  }
+  // Store the filter states.
+  filter_state[0] = tmp32_1;
+  filter_state[1] = tmp32_2;
+}
+
+// Inserts |feature_value| into |low_value_vector|, if it is one of the 16
+// smallest values the last 100 frames. Then calculates and returns the median
+// of the five smallest values.
+int16_t WebRtcVad_FindMinimum(VadInstT* self,
+                              int16_t feature_value,
+                              int channel) {
+  int i = 0, j = 0;
+  int position = -1;
+  // Offset to beginning of the 16 minimum values in memory.
+  const int offset = (channel << 4);
+  int16_t current_median = 1600;
+  int16_t alpha = 0;
+  int32_t tmp32 = 0;
+  // Pointer to memory for the 16 minimum values and the age of each value of
+  // the |channel|.
+  int16_t* age = &self->index_vector[offset];
+  int16_t* smallest_values = &self->low_value_vector[offset];
+
+  assert(channel < kNumChannels);
+
+  // Each value in |smallest_values| is getting 1 loop older. Update |age|, and
+  // remove old values.
+  for (i = 0; i < 16; i++) {
+    if (age[i] != 100) {
+      age[i]++;
+    } else {
+      // Too old value. Remove from memory and shift larger values downwards.
+      for (j = i; j < 16; j++) {
+        smallest_values[j] = smallest_values[j + 1];
+        age[j] = age[j + 1];
+      }
+      age[15] = 101;
+      smallest_values[15] = 10000;
+    }
+  }
+
+  // Check if |feature_value| is smaller than any of the values in
+  // |smallest_values|. If so, find the |position| where to insert the new value
+  // (|feature_value|).
+  if (feature_value < smallest_values[7]) {
+    if (feature_value < smallest_values[3]) {
+      if (feature_value < smallest_values[1]) {
+        if (feature_value < smallest_values[0]) {
+          position = 0;
+        } else {
+          position = 1;
+        }
+      } else if (feature_value < smallest_values[2]) {
+        position = 2;
+      } else {
+        position = 3;
+      }
+    } else if (feature_value < smallest_values[5]) {
+      if (feature_value < smallest_values[4]) {
+        position = 4;
+      } else {
+        position = 5;
+      }
+    } else if (feature_value < smallest_values[6]) {
+      position = 6;
+    } else {
+      position = 7;
+    }
+  } else if (feature_value < smallest_values[15]) {
+    if (feature_value < smallest_values[11]) {
+      if (feature_value < smallest_values[9]) {
+        if (feature_value < smallest_values[8]) {
+          position = 8;
+        } else {
+          position = 9;
+        }
+      } else if (feature_value < smallest_values[10]) {
+        position = 10;
+      } else {
+        position = 11;
+      }
+    } else if (feature_value < smallest_values[13]) {
+      if (feature_value < smallest_values[12]) {
+        position = 12;
+      } else {
+        position = 13;
+      }
+    } else if (feature_value < smallest_values[14]) {
+      position = 14;
+    } else {
+      position = 15;
+    }
+  }
+
+  // If we have detected a new small value, insert it at the correct position
+  // and shift larger values up.
+  if (position > -1) {
+    for (i = 15; i > position; i--) {
+      smallest_values[i] = smallest_values[i - 1];
+      age[i] = age[i - 1];
+    }
+    smallest_values[position] = feature_value;
+    age[position] = 1;
+  }
+
+  // Get |current_median|.
+  if (self->frame_counter > 2) {
+    current_median = smallest_values[2];
+  } else if (self->frame_counter > 0) {
+    current_median = smallest_values[0];
+  }
+
+  // Smooth the median value.
+  if (self->frame_counter > 0) {
+    if (current_median < self->mean_value[channel]) {
+      alpha = kSmoothingDown;  // 0.2 in Q15.
+    } else {
+      alpha = kSmoothingUp;  // 0.99 in Q15.
+    }
+  }
+  tmp32 = WEBRTC_SPL_MUL_16_16(alpha + 1, self->mean_value[channel]);
+  tmp32 += WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_WORD16_MAX - alpha, current_median);
+  tmp32 += 16384;
+  self->mean_value[channel] = (int16_t) (tmp32 >> 15);
+
+  return self->mean_value[channel];
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_sp.h b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_sp.h
new file mode 100644
index 0000000..6c05ea3
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/vad_sp.h
@@ -0,0 +1,56 @@
+/*
+ *  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 specific signal processing tools used in vad_core.c.
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
+
+#include "vad_core.h"
+#include "../include/typedefs.h"
+
+// Downsamples the signal by a factor 2, eg. 32->16 or 16->8.
+//
+// Inputs:
+//      - signal_in     : Input signal.
+//      - in_length     : Length of input signal in samples.
+//
+// Input & Output:
+//      - filter_state  : Current filter states of the two all-pass filters. The
+//                        |filter_state| is updated after all samples have been
+//                        processed.
+//
+// Output:
+//      - signal_out    : Downsampled signal (of length |in_length| / 2).
+void WebRtcVad_Downsampling(int16_t* signal_in,
+                            int16_t* signal_out,
+                            int32_t* filter_state,
+                            int in_length);
+
+// Updates and returns the smoothed feature minimum. As minimum we use the
+// median of the five smallest feature values in a 100 frames long window.
+// As long as |handle->frame_counter| is zero, that is, we haven't received any
+// "valid" data, FindMinimum() outputs the default value of 1600.
+//
+// Inputs:
+//      - feature_value : New feature value to update with.
+//      - channel       : Channel number.
+//
+// Input & Output:
+//      - handle        : State information of the VAD.
+//
+// Returns:
+//                      : Smoothed minimum value for a moving window.
+int16_t WebRtcVad_FindMinimum(VadInstT* handle,
+                              int16_t feature_value,
+                              int channel);
+
+#endif  // WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/webrtc_vad.c b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/webrtc_vad.c
new file mode 100644
index 0000000..d3e87d7
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc/common_audio/vad/webrtc_vad.c
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2012 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 "include/webrtc_vad.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "../signal_processing/include/signal_processing_library.h"
+#include "vad_core.h"
+#include "../include/typedefs.h"
+
+static const int kInitCheck = 42;
+static const int kValidRates[] = { 8000, 16000, 32000, 48000 };
+static const size_t kRatesSize = sizeof(kValidRates) / sizeof(*kValidRates);
+static const int kMaxFrameLengthMs = 30;
+
+int WebRtcVad_Create(VadInst** handle) {
+  VadInstT* self = NULL;
+
+  if (handle == NULL) {
+    return -1;
+  }
+
+  *handle = NULL;
+  self = (VadInstT*) malloc(sizeof(VadInstT));
+  *handle = (VadInst*) self;
+
+  if (self == NULL) {
+    return -1;
+  }
+
+  WebRtcSpl_Init();
+
+  self->init_flag = 0;
+
+  return 0;
+}
+
+int WebRtcVad_Free(VadInst* handle) {
+  if (handle == NULL) {
+    return -1;
+  }
+
+  free(handle);
+
+  return 0;
+}
+
+// TODO(bjornv): Move WebRtcVad_InitCore() code here.
+int WebRtcVad_Init(VadInst* handle) {
+  // Initialize the core VAD component.
+  return WebRtcVad_InitCore((VadInstT*) handle);
+}
+
+// TODO(bjornv): Move WebRtcVad_set_mode_core() code here.
+int WebRtcVad_set_mode(VadInst* handle, int mode) {
+  VadInstT* self = (VadInstT*) handle;
+
+  if (handle == NULL) {
+    return -1;
+  }
+  if (self->init_flag != kInitCheck) {
+    return -1;
+  }
+
+  return WebRtcVad_set_mode_core(self, mode);
+}
+
+int WebRtcVad_Process(VadInst* handle, int fs, int16_t* audio_frame,
+                      int frame_length) {
+  int vad = -1;
+  VadInstT* self = (VadInstT*) handle;
+
+  if (handle == NULL) {
+    return -1;
+  }
+
+  if (self->init_flag != kInitCheck) {
+    return -1;
+  }
+  if (audio_frame == NULL) {
+    return -1;
+  }
+  if (WebRtcVad_ValidRateAndFrameLength(fs, frame_length) != 0) {
+    return -1;
+  }
+
+  if (fs == 48000) {
+      vad = WebRtcVad_CalcVad48khz(self, audio_frame, frame_length);
+  } else if (fs == 32000) {
+    vad = WebRtcVad_CalcVad32khz(self, audio_frame, frame_length);
+  } else if (fs == 16000) {
+    vad = WebRtcVad_CalcVad16khz(self, audio_frame, frame_length);
+  } else if (fs == 8000) {
+    vad = WebRtcVad_CalcVad8khz(self, audio_frame, frame_length);
+  }
+
+  if (vad > 0) {
+    vad = 1;
+  }
+  return vad;
+}
+
+int WebRtcVad_ValidRateAndFrameLength(int rate, int frame_length) {
+  int return_value = -1;
+  size_t i;
+  int valid_length_ms;
+  int valid_length;
+
+  // We only allow 10, 20 or 30 ms frames. Loop through valid frame rates and
+  // see if we have a matching pair.
+  for (i = 0; i < kRatesSize; i++) {
+    if (kValidRates[i] == rate) {
+      for (valid_length_ms = 10; valid_length_ms <= kMaxFrameLengthMs;
+          valid_length_ms += 10) {
+        valid_length = (kValidRates[i] / 1000 * valid_length_ms);
+        if (frame_length == valid_length) {
+          return_value = 0;
+          break;
+        }
+      }
+      break;
+    }
+  }
+
+  return return_value;
+}
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc_jni.c b/mozillaspeechlibrary/src/main/cpp/webrtc_jni.c
new file mode 100644
index 0000000..cae068d
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc_jni.c
@@ -0,0 +1,43 @@
+#include "webrtc/common_audio/vad/include/webrtc_vad.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include <stdlib.h>
+#include "webrtc/common_audio/include/typedefs.h"
+#include <jni.h>
+
+#define AGGRESSIVENESS 3
+#define BUFFER_SIZE 512
+VadInst* internalHandle;
+int resultVad;
+
+JNIEXPORT jint JNICALL Java_com_mozilla_speechlibrary_Vad_start (JNIEnv * env, jobject obj){
+    short buffer[BUFFER_SIZE];
+    int ret_state = 0;
+    internalHandle = NULL;
+
+    ret_state = WebRtcVad_Create(&internalHandle);
+    if (ret_state == -1) return -1;
+    ret_state =  WebRtcVad_Init(internalHandle);
+    if (ret_state == -1) return -2;
+    ret_state = WebRtcVad_set_mode(internalHandle, AGGRESSIVENESS);
+    if (ret_state == -1) return -3;
+    return ret_state;
+}
+
+
+JNIEXPORT jint JNICALL Java_com_mozilla_speechlibrary_Vad_stop(JNIEnv * env, jobject object) {
+    WebRtcVad_Free(internalHandle);
+}
+
+JNIEXPORT jint JNICALL Java_com_mozilla_speechlibrary_Vad_isSilence(JNIEnv * env, jobject object) {
+    return resultVad;
+}
+  
+
+JNIEXPORT jint JNICALL Java_com_mozilla_speechlibrary_Vad_feed(JNIEnv * env, jobject object, jshortArray bytes, jint size) {
+    jshort * arrayElements = (*env)->GetShortArrayElements(env, bytes, 0);
+    resultVad = WebRtcVad_Process(internalHandle, 16000, arrayElements, size);
+    (*env)->ReleaseShortArrayElements( env, bytes, arrayElements, 0 );
+    return resultVad;
+}
+
+
diff --git a/mozillaspeechlibrary/src/main/cpp/webrtc_vad.h b/mozillaspeechlibrary/src/main/cpp/webrtc_vad.h
new file mode 100644
index 0000000..21394e5
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/cpp/webrtc_vad.h
@@ -0,0 +1,159 @@
+/*
+ *  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_
diff --git a/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Encoder.java b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Encoder.java
new file mode 100644
index 0000000..7e49a10
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Encoder.java
@@ -0,0 +1,9 @@
+package com.github.axet.audiolibrary.encoders;
+
+public interface Encoder {
+
+    void encode(short[] buf, int pos, int len);
+
+    void close();
+
+}
diff --git a/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/EncoderInfo.java b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/EncoderInfo.java
new file mode 100644
index 0000000..040005f
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/EncoderInfo.java
@@ -0,0 +1,14 @@
+package com.github.axet.audiolibrary.encoders;
+
+public class EncoderInfo {
+    public int channels;
+    public int hz;
+    public int bps;
+
+    public EncoderInfo(int channels, int sampleRate, int bps) {
+        this.channels = channels;
+        this.hz = sampleRate;
+        this.bps = bps;
+    }
+
+}
diff --git a/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Factory.java b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Factory.java
new file mode 100644
index 0000000..5661fdd
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Factory.java
@@ -0,0 +1,46 @@
+package com.github.axet.audiolibrary.encoders;
+
+import android.content.Context;
+import android.media.AudioFormat;
+import android.os.Build;
+
+import java.io.ByteArrayOutputStream;
+import java.io.FileDescriptor;
+
+public class Factory {
+
+    public static int getBitrate(int hz) {
+        if (hz < 16000) {
+            return 32000;
+        } else if (hz < 44100) {
+            return 64000;
+        } else {
+            return 128000;
+        }
+    }
+
+    public static Encoder getEncoder(Context context, String ext, EncoderInfo info, ByteArrayOutputStream out) {
+        if (ext.equals(FormatOPUS.EXT)) {
+            if (Build.VERSION.SDK_INT >= 23) { // Android 6.0 (has ogg/opus support) https://en.wikipedia.org/wiki/Opus_(audio_format)
+                return new FormatOPUS_OGG(context, info, out); // android6+ supports ogg/opus
+            }
+        }
+        return null;
+    }
+
+    public static long getEncoderRate(String ext, int rate) {
+        if (ext.equals(FormatOPUS.EXT)) {
+            long y1 = 202787; // one minute sample 16000Hz
+            long x1 = 16000; // at 16000
+            long y2 = 319120; // one minute sample
+            long x2 = 44000; // at 44000
+            long x = rate;
+            long y = (x - x1) * (y2 - y1) / (x2 - x1) + y1;
+            return y / 60;
+        }
+
+        // default raw
+        int c = Sound.DEFAULT_AUDIOFORMAT == AudioFormat.ENCODING_PCM_16BIT ? 2 : 1;
+        return c * rate;
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/FormatOPUS.java b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/FormatOPUS.java
new file mode 100755
index 0000000..b3ec52a
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/FormatOPUS.java
@@ -0,0 +1,173 @@
+package com.github.axet.audiolibrary.encoders;
+
+import android.annotation.TargetApi;
+import android.content.Context;
+import com.github.axet.opusjni.Config;
+import com.github.axet.opusjni.Opus;
+import java.io.ByteArrayOutputStream;
+import java.nio.ByteBuffer;
+import java.nio.ShortBuffer;
+
+@TargetApi(21)
+public class FormatOPUS implements Encoder {
+    public static final String TAG = FormatOPUS.class.getSimpleName();
+
+    public static final String EXT = "opus";
+
+    public static final int SHORT_BYTES = Short.SIZE / Byte.SIZE;
+
+    EncoderInfo info;
+    Opus opus;
+    long NumSamples;
+    ShortBuffer left;
+    int frameSize = 960; // default 20ms
+    int hz;
+    Resample resample;
+
+    public static void natives(Context context) {
+        if (Config.natives) {
+            Natives.loadLibraries(context, "opus", "opusjni");
+            Config.natives = false;
+        }
+    }
+
+    public static boolean supported(Context context) {
+        try {
+            FormatOPUS.natives(context);
+            Opus v = new Opus();
+            return true;
+        } catch (NoClassDefFoundError | ExceptionInInitializerError | UnsatisfiedLinkError e) {
+            return false;
+        }
+    }
+
+    public static int getBitrate(int hz) { // https://wiki.xiph.org/index.php?title=Opus_Recommended_Settings
+        if (hz < 16000) {
+            return 16000; // 0 - 16Hz
+        } else if (hz < 44100) {
+            return 24000; // 16 - 44Hz
+        } else {
+            return 32000; // 48Hz
+        }
+    }
+
+    public static int match(int hz) { // opus supports only selected Hz's
+        int[] hh = new int[]{
+                8000,
+                12000,
+                16000,
+                24000,
+                48000,
+        };
+        int i = Integer.MAX_VALUE;
+        int r = 0;
+        for (int h : hh) {
+            int d = Math.abs(hz - h);
+            if (d <= i) { // higher is better
+                i = d;
+                r = h;
+            }
+        }
+        return r;
+    }
+
+    public FormatOPUS(Context context, EncoderInfo info, ByteArrayOutputStream out) {
+        natives(context);
+        create(info, out);
+    }
+
+    public void create(final EncoderInfo info, ByteArrayOutputStream out) {
+        this.info = info;
+        this.hz = match(info.hz);
+
+        if (hz != info.hz)
+            resample = new Resample(info.hz, info.channels, hz);
+
+        opus = new Opus();
+        opus.open(info.channels, hz, getBitrate(info.hz));
+    }
+
+    @Override
+    public void encode(short[] buf, int pos, int len) {
+        if (resample != null) {
+            resample.write(buf, pos, len);
+            resample();
+            return;
+        }
+        encode2(buf, pos, len);
+    }
+
+    void encode2(short[] buf, int pos, int len) {
+        if (left != null) {
+            ShortBuffer ss = ShortBuffer.allocate(left.position() + len);
+            left.flip();
+            ss.put(left);
+            ss.put(buf, pos, len);
+            buf = ss.array();
+            pos = 0;
+            len = ss.position();
+        }
+        if (frameSize == 0) {
+            if (len < 240) {
+                frameSize = 120;
+            } else if (len < 480) {
+                frameSize = 240;
+            } else if (len < 960) {
+                frameSize = 480;
+            } else if (len < 1920) {
+                frameSize = 960;
+            } else if (len < 2880) {
+                frameSize = 1920;
+            } else {
+                frameSize = 2880;
+            }
+        }
+        int frameSizeStereo = frameSize * info.channels;
+        int lenEncode = len / frameSizeStereo * frameSizeStereo;
+        int end = pos + lenEncode;
+        for (int p = pos; p < end; p += frameSizeStereo) {
+            byte[] bb = opus.encode(buf, p, frameSizeStereo);
+            encode(ByteBuffer.wrap(bb), frameSize);
+            NumSamples += frameSizeStereo / info.channels;
+        }
+        int diff = len - lenEncode;
+        if (diff > 0) {
+            left = ShortBuffer.allocate(diff);
+            left.put(buf, end, diff);
+        } else {
+            left = null;
+        }
+    }
+
+    void resample() {
+        ByteBuffer bb;
+        while ((bb = resample.read()) != null) {
+            int len = bb.position() / SHORT_BYTES;
+            short[] b = new short[len];
+            bb.flip();
+            bb.asShortBuffer().get(b, 0, len);
+            encode2(b, 0, len);
+        }
+    }
+
+    void encode(ByteBuffer bb, long dur) {
+    }
+
+    public void close() {
+        if (resample != null) {
+            resample.end();
+            resample();
+            resample.close();
+            resample = null;
+        }
+        opus.close();
+    }
+
+    long getCurrentTimeStamp() {
+        return NumSamples * 1000 / info.hz;
+    }
+
+    public EncoderInfo getInfo() {
+        return info;
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/FormatOPUS_OGG.java b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/FormatOPUS_OGG.java
new file mode 100755
index 0000000..d099f3f
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/FormatOPUS_OGG.java
@@ -0,0 +1,85 @@
+package com.github.axet.audiolibrary.encoders;
+
+import android.annotation.TargetApi;
+import android.content.Context;
+import org.gagravarr.ogg.OggFile;
+import org.gagravarr.ogg.OggPacketWriter;
+import org.gagravarr.opus.OpusAudioData;
+import org.gagravarr.opus.OpusInfo;
+import org.gagravarr.opus.OpusTags;
+import java.io.ByteArrayOutputStream;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+
+// https://wiki.xiph.org/OggOpus
+@TargetApi(23)
+public class FormatOPUS_OGG extends FormatOPUS {
+    public static final String TAG = FormatOPUS_OGG.class.getSimpleName();
+
+    OggFile file;
+    OggPacketWriter writer;
+    long lastGranule = 0;
+
+    public FormatOPUS_OGG(Context context, EncoderInfo info, ByteArrayOutputStream out) {
+        super(context, info, out);
+    }
+
+    @Override
+    public void create(final EncoderInfo info, ByteArrayOutputStream out) {
+        super.create(info, out);
+        try {
+            OpusInfo oinfo = new OpusInfo();
+            oinfo.setNumChannels(info.channels);
+            oinfo.setOutputGain(0);
+            oinfo.setPreSkip(0);
+            oinfo.setSampleRate(info.hz);
+            OpusTags otags = new OpusTags();
+            file = new OggFile(out);
+            writer = file.getPacketWriter();
+            writer.bufferPacket(oinfo.write());
+            writer.flush();
+            writer.bufferPacket(otags.write());
+            writer.flush();
+        } catch (IOException e) {
+            throw new RuntimeException(e);
+        }
+    }
+
+    @Override
+    void encode(ByteBuffer bb, long dur) {
+        OpusAudioData frame = new OpusAudioData(bb.array());
+        long end = NumSamples + dur;
+        long gr = OpusAudioData.OPUS_GRANULE_RATE * end / info.hz; // Ogg gr always at 48000hz
+        frame.setGranulePosition(gr);
+        try {
+            if (frame.getGranulePosition() >= 0 && lastGranule != frame.getGranulePosition()) {
+                writer.flush();
+                lastGranule = frame.getGranulePosition();
+                writer.setGranulePosition(lastGranule);
+            }
+            writer.bufferPacket(frame.write());
+            if (writer.getSizePendingFlush() > 16384) {
+                writer.flush();
+            }
+        } catch (IOException e) {
+            throw new RuntimeException(e);
+        }
+    }
+
+    @Override
+    public void close() {
+        super.close();
+        try {
+            if (writer != null) {
+                writer.close();
+                writer = null;
+            }
+            if (file != null) {
+                file.close();
+                file = null;
+            }
+        } catch (IOException e) {
+            throw new RuntimeException(e);
+        }
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Natives.java b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Natives.java
new file mode 100644
index 0000000..3493a89
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Natives.java
@@ -0,0 +1,94 @@
+package com.github.axet.audiolibrary.encoders;
+
+import android.content.Context;
+import android.os.Build;
+
+import java.io.File;
+import java.util.ArrayList;
+import java.util.Collections;
+import java.util.Comparator;
+import java.util.List;
+
+public class Natives {
+
+    public static String ARCH = Build.CPU_ABI;
+
+    public static class ArchFirst implements Comparator<File> {
+        @Override
+        public int compare(File o1, File o2) {
+            String p1 = o1.getPath();
+            String p2 = o2.getPath();
+            boolean b1 = p1.contains(ARCH);
+            boolean b2 = p2.contains(ARCH);
+            if (b1 && b2)
+                return p1.compareTo(p2);
+            if (b1)
+                return -1;
+            if (b2)
+                return 1;
+            return p1.compareTo(p2);
+        }
+    }
+
+    public static void loadLibraries(Context context, String... libs) {
+        try {
+            for (String l : libs) {
+                System.loadLibrary(l); // API16 failed to find dependencies
+            }
+        } catch (ExceptionInInitializerError | UnsatisfiedLinkError e) { // API15 crash
+            for (String l : libs) {
+                Natives.loadLibrary(context, l);
+            }
+        }
+    }
+
+    /**
+     * API15 crash while loading wrong arch native libraries. We need to find and load them manually.
+     * <p>
+     * Caused by: java.lang.UnsatisfiedLinkError: Cannot load library: reloc_library[1286]:  1823 cannot locate '__aeabi_idiv0'...
+     * at java.lang.Runtime.loadLibrary(Runtime.java:370)
+     * at java.lang.System.loadLibrary(System.java:535)
+     */
+    public static void loadLibrary(final Context context, String libname) {
+        String file = search(context, System.mapLibraryName(libname));
+        if (file == null)
+            throw new UnsatisfiedLinkError("file not found: " + libname);
+        System.load(file);
+    }
+
+    public static String search(Context context, String filename) {
+        String dir = context.getApplicationInfo().nativeLibraryDir;
+        if (dir.endsWith(ARCH)) {
+            File f = new File(dir);
+            f = f.getParentFile();
+            String lib = search(f, filename);
+            if (lib != null)
+                return lib;
+        }
+        return search(new File(dir), filename);
+    }
+
+    public static String search(File f, String filename) {
+        List<File> ff = list(f, filename);
+        Collections.sort(ff, new ArchFirst());
+        if (ff.size() == 0)
+            return null;
+        return ff.get(0).getAbsolutePath();
+    }
+
+    public static ArrayList<File> list(File f, String filename) {
+        ArrayList<File> ff = new ArrayList<>();
+        File[] aa = f.listFiles();
+        if (aa != null) {
+            for (File a : aa) {
+                if (a.isDirectory()) {
+                    ArrayList<File> mm = list(a, filename);
+                    ff.addAll(mm);
+                }
+                if (a.getName().equals(filename))
+                    ff.add(a);
+            }
+        }
+        return ff;
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Resample.java b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Resample.java
new file mode 100644
index 0000000..521e16b
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Resample.java
@@ -0,0 +1,104 @@
+package com.github.axet.audiolibrary.encoders;
+
+import android.media.AudioFormat;
+import android.util.Log;
+import java.io.IOException;
+import java.io.PipedInputStream;
+import java.io.PipedOutputStream;
+import java.nio.ByteBuffer;
+import java.nio.ByteOrder;
+import vavi.sound.pcm.resampling.ssrc.SSRC;
+
+public class Resample {
+    public static final String TAG = Resample.class.getSimpleName();
+
+    public static final ByteOrder ORDER = ByteOrder.LITTLE_ENDIAN;
+    public static final int SHORT_BYTES = Short.SIZE / Byte.SIZE;
+    public static final int PIPE_SIZE = 100 * 1024;
+
+    Thread thread;
+    PipedOutputStream os;
+    PipedInputStream is;
+    RuntimeException delayed;
+
+    public Resample(final int sampleRate, final int channels, final int hz) {
+        try {
+            this.os = new PipedOutputStream();
+            this.is = new PipedInputStream(PIPE_SIZE);
+            final PipedInputStream is = new PipedInputStream(this.os);
+            final PipedOutputStream os = new PipedOutputStream(this.is);
+            final int c = Sound.DEFAULT_AUDIOFORMAT == AudioFormat.ENCODING_PCM_16BIT ? 2 : 1;
+            thread = new Thread(new Runnable() {
+                @Override
+                public void run() {
+                    try {
+                        SSRC ssrc = new SSRC(is, os, sampleRate, hz, c, c, channels, Integer.MAX_VALUE, 0, 0, true);
+                    } catch (RuntimeException e) {
+                        Log.d(TAG, "SSRC failed", e);
+                        delayed = e;
+                    } catch (IOException e) {
+                        Log.d(TAG, "SSRC failed", e);
+                        delayed = new RuntimeException(e);
+                    }
+                }
+            }, "SSRC");
+            thread.start();
+        } catch (IOException e) {
+            throw new RuntimeException(e);
+        }
+    }
+
+    public void end() {
+        if (delayed != null)
+            throw delayed;
+        try {
+            is.close();
+        } catch (IOException e) {
+            throw new RuntimeException(e);
+        }
+    }
+
+    public void write(short[] buf, int pos, int len) {
+        if (delayed != null)
+            throw delayed;
+        try {
+            ByteBuffer bb = ByteBuffer.allocate(len * SHORT_BYTES);
+            bb.order(ORDER);
+            bb.asShortBuffer().put(buf, pos, len);
+            os.write(bb.array());
+            os.flush();
+        } catch (IOException e) {
+            throw new RuntimeException(e);
+        }
+    }
+
+    public ByteBuffer read() {
+        if (delayed != null)
+            throw delayed;
+        try {
+            int blen = is.available();
+            if (blen <= 0)
+                return null;
+            byte[] b = new byte[blen];
+            int read = is.read(b);
+            ByteBuffer bb = ByteBuffer.allocate(read);
+            bb.order(ORDER);
+            bb.put(b, 0, read);
+            return bb;
+        } catch (IOException e) {
+            throw new RuntimeException(e);
+        }
+    }
+
+    public void close() {
+        if (thread != null) {
+            thread.interrupt();
+            try {
+                thread.join();
+            } catch (InterruptedException e) {
+                Thread.currentThread().interrupt();
+            }
+            thread = null;
+        }
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Sound.java b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Sound.java
new file mode 100644
index 0000000..b91a2e3
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/github/axet/audiolibrary/encoders/Sound.java
@@ -0,0 +1,74 @@
+package com.github.axet.audiolibrary.encoders;
+
+import android.media.AudioFormat;
+import android.media.AudioRecord;
+import android.media.MediaRecorder;
+import org.apache.commons.math3.complex.Complex;
+import org.apache.commons.math3.transform.DftNormalization;
+import org.apache.commons.math3.transform.FastFourierTransformer;
+import org.apache.commons.math3.transform.TransformType;
+
+public class Sound {
+    public static final int DEFAULT_AUDIOFORMAT = AudioFormat.ENCODING_PCM_16BIT;
+
+    public static AudioRecord getAudioRecord(int aNumChannels, int aSampleRate){
+        int minBufSize = AudioRecord.getMinBufferSize(aSampleRate,
+                aNumChannels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO,
+                AudioFormat.ENCODING_PCM_16BIT);
+
+        // initialize audio recorder
+        AudioRecord recorder = new AudioRecord(MediaRecorder.AudioSource.MIC,
+                aSampleRate,
+                aNumChannels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO,
+                AudioFormat.ENCODING_PCM_16BIT,
+                minBufSize);
+
+        return recorder;
+
+    }
+
+    public static double[] fft(short[] buffer, int offset, int len) {
+        int len2 = (int) Math.pow(2, Math.ceil(Math.log(len) / Math.log(2)));
+
+        final double[][] dataRI = new double[][]{
+                new double[len2], new double[len2]
+        };
+
+        double[] dataR = dataRI[0];
+        double[] dataI = dataRI[1];
+
+        double powerInput = 0;
+        for (int i = 0; i < len; i++) {
+            dataR[i] = buffer[offset + i] / (float) 0x7fff;
+            powerInput += dataR[i] * dataR[i];
+        }
+        powerInput = Math.sqrt(powerInput / len);
+
+        FastFourierTransformer.transformInPlace(dataRI, DftNormalization.STANDARD, TransformType.FORWARD);
+
+        double[] data = new double[len2 / 2];
+
+        data[0] = 10 * Math.log10(Math.pow(new Complex(dataR[0], dataI[0]).abs() / len2, 2));
+
+        double powerOutput = 0;
+        for (int i = 1; i < data.length; i++) {
+            Complex c = new Complex(dataR[i], dataI[i]);
+            double p = c.abs();
+            p = p / len2;
+            p = p * p;
+            p = p * 2;
+            double dB = 10 * Math.log10(p);
+
+            powerOutput += p;
+            data[i] = dB;
+        }
+        powerOutput = Math.sqrt(powerOutput);
+
+//        if(powerInput != powerOutput) {
+//            throw new RuntimeException("in " + powerInput + " out " + powerOutput);
+//        }
+
+        return data;
+    }
+
+}
diff --git a/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/ISpeechRecognitionListener.java b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/ISpeechRecognitionListener.java
new file mode 100644
index 0000000..482a26e
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/ISpeechRecognitionListener.java
@@ -0,0 +1,5 @@
+package com.mozilla.speechlibrary;
+
+public interface ISpeechRecognitionListener {
+    void onSpeechStatusChanged(MozillaSpeechService.SpeechState aState, Object aPayload);
+}
\ No newline at end of file
diff --git a/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/MozillaSpeechService.java b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/MozillaSpeechService.java
new file mode 100644
index 0000000..d8dd851
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/MozillaSpeechService.java
@@ -0,0 +1,80 @@
+package com.mozilla.speechlibrary;
+
+import android.app.Activity;
+import android.content.Context;
+import android.util.Log;
+
+import java.util.ArrayList;
+
+public class MozillaSpeechService {
+
+    protected static final String TAG = "MozillaSpeech";
+    private final int SAMPLERATE = 16000;
+    private final int CHANNELS = 1;
+    private ArrayList<ISpeechRecognitionListener> mListeners;
+    private Context mContext;
+    private boolean isIdle = true;
+
+    public enum SpeechState
+    {
+        DECODING, MIC_ACTIVITY, STT_RESULT, START_LISTEN,
+        NO_VOICE, CANCELED, ERROR
+    }
+
+    private static final MozillaSpeechService ourInstance = new MozillaSpeechService();
+    private SpeechRecognition mSpeechRecognition;
+    private SpeechState mState;
+    private Vad mVad;
+
+    public static MozillaSpeechService getInstance() {
+        return ourInstance;
+    }
+
+    private MozillaSpeechService() {
+    }
+
+    public void start(Context aContext) {
+
+        try {
+            if (!isIdle) {
+                notifyListeners(SpeechState.ERROR, "Recognition already In progress");
+            } else {
+                mVad = new Vad();
+                int retVal = mVad.start();
+                this.mContext = aContext;
+                if (retVal < 0) {
+                    notifyListeners(SpeechState.ERROR, "Error Initializing VAD " + String.valueOf(retVal));
+                } else {
+                    this.mSpeechRecognition = new SpeechRecognition(SAMPLERATE, CHANNELS, mVad, aContext, this);
+                    Thread audio_thread = new Thread(this.mSpeechRecognition);
+                    audio_thread.start();
+                    isIdle = false;
+                }
+            }
+        } catch (Exception exc) {
+            notifyListeners(SpeechState.ERROR, "General error loading the module.");
+        }
+    }
+
+    public void addListener(ISpeechRecognitionListener aListener) {
+        if (mListeners == null) {
+            mListeners = new ArrayList<>();
+        }
+        mListeners.add(aListener);
+    }
+
+    public void notifyListeners(MozillaSpeechService.SpeechState aState, Object aPayload) {
+        mState = aState;
+        for (ISpeechRecognitionListener listener : mListeners) {
+            listener.onSpeechStatusChanged(aState, aPayload);
+            if (aState == SpeechState.STT_RESULT || aState == SpeechState.ERROR
+                    || aState == SpeechState.NO_VOICE || aState == SpeechState.CANCELED) {
+                isIdle = true;
+            }
+        }
+    }
+
+    public void cancel() {
+        this.mSpeechRecognition.cancel();
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/Networking.java b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/Networking.java
new file mode 100644
index 0000000..77b737d
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/Networking.java
@@ -0,0 +1,94 @@
+package com.mozilla.speechlibrary;
+
+import android.content.Context;
+import android.os.Looper;
+import android.util.Log;
+import com.loopj.android.http.AsyncHttpResponseHandler;
+import com.loopj.android.http.SyncHttpClient;
+import org.json.JSONArray;
+import org.json.JSONObject;
+import cz.msebera.android.httpclient.Header;
+import cz.msebera.android.httpclient.entity.ByteArrayEntity;
+import java.io.ByteArrayOutputStream;
+import static com.mozilla.speechlibrary.MozillaSpeechService.TAG;
+
+public class Networking {
+
+    final String STT_ENDPOINT = "https://speaktome-2.services.mozilla.com/";
+    MozillaSpeechService mSpeechService;
+    public boolean cancelled;
+    protected Context mContext;
+
+    public Networking(MozillaSpeechService aSpeechService) {
+        this.mSpeechService = aSpeechService;
+    }
+
+    protected void doSTT(final ByteArrayOutputStream baos) {
+
+        if (cancelled) {
+            mSpeechService.notifyListeners(MozillaSpeechService.SpeechState.CANCELED, null);
+            return;
+        }
+
+        try {
+            Looper.prepare();
+            ByteArrayEntity byteArrayEntity = new ByteArrayEntity(baos.toByteArray());
+            SyncHttpClient client = new SyncHttpClient();
+            client.post(mContext,STT_ENDPOINT, byteArrayEntity, "audio/3gpp",
+                new AsyncHttpResponseHandler() {
+
+                    @Override
+                    public void onStart() {
+                        // called before request is started
+                        mSpeechService.notifyListeners(MozillaSpeechService.SpeechState.DECODING, null);
+                    }
+
+                    @Override
+                    public void onSuccess(int statusCode, Header[] headers, byte[] response) {
+                        // Implement cancelation
+                        if (cancelled) {
+                            return;
+                        }
+
+                        // called when response HTTP status is "200 OK"
+                        String json = new String(response);
+                        try {
+                            JSONObject reader = new JSONObject(json);
+                            JSONArray results = reader.getJSONArray("data");
+                            final String transcription = results.getJSONObject(0).getString("text");
+                            final String confidence = results.getJSONObject(0).getString("confidence");
+                            STTResult sttResult = new STTResult(transcription, Float.parseFloat(confidence));
+                            mSpeechService.notifyListeners(MozillaSpeechService.SpeechState.STT_RESULT, sttResult);
+                        } catch (Exception exc) {
+                            String error = String.format("Error parsing results: %s", exc.getMessage());
+                            mSpeechService.notifyListeners(MozillaSpeechService.SpeechState.ERROR, error);
+                        }
+                    }
+
+                    @Override
+                    public void onFailure(int statusCode, Header[] headers, byte[] errorResponse, Throwable e) {
+                        // called when response HTTP status is "4XX" (eg. 401, 403, 404)
+                        String error = String.format("Network Error: %s (%s)", errorResponse == null ? "General Error" : new String(errorResponse), String.valueOf(statusCode));
+                        mSpeechService.notifyListeners(MozillaSpeechService.SpeechState.ERROR, error);
+                    }
+
+                    @Override
+                    public void onRetry(int retryNo) {
+                        // called when request is retried
+                        String error = String.format("Network Error: Retrying %s", String.valueOf(retryNo));
+                        mSpeechService.notifyListeners(MozillaSpeechService.SpeechState.ERROR, error);
+                    }
+                }
+             );
+        }
+        catch(Exception e)
+        {
+            e.printStackTrace();
+        }
+    }
+
+    public void cancel() {
+        cancelled = true;
+    }
+}
+
diff --git a/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/STTResult.java b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/STTResult.java
new file mode 100644
index 0000000..b40f3a7
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/STTResult.java
@@ -0,0 +1,12 @@
+package com.mozilla.speechlibrary;
+
+public class STTResult {
+
+    public String mTranscription;
+    public float mConfidence;
+
+    public STTResult(String aTranscription, float aConfidence) {
+        this.mTranscription = aTranscription;
+        this.mConfidence = aConfidence;
+    }
+}
\ No newline at end of file
diff --git a/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/SpeechRecognition.java b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/SpeechRecognition.java
new file mode 100644
index 0000000..88bad3a
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/SpeechRecognition.java
@@ -0,0 +1,140 @@
+package com.mozilla.speechlibrary;
+
+import android.content.Context;
+import android.media.AudioRecord;
+import android.os.Process;
+import com.github.axet.audiolibrary.encoders.Encoder;
+import com.github.axet.audiolibrary.encoders.EncoderInfo;
+import com.github.axet.audiolibrary.encoders.Factory;
+import com.github.axet.audiolibrary.encoders.FormatOPUS;
+import com.github.axet.audiolibrary.encoders.Sound;
+import java.io.ByteArrayOutputStream;
+import java.util.Arrays;
+
+class SpeechRecognition implements Runnable {
+
+    Vad mVad;
+    short[] mBuftemp;
+    ByteArrayOutputStream baos ;
+    int mMinimumVoice = 250;
+    int mMaximumSilence = 1500;
+    int mUpperLimit = 10;
+    static final int FRAME_SIZE = 160;
+    boolean done;
+    boolean cancelled;
+    Context mContext;
+    int mSampleRate;
+    int mChannels;
+    MozillaSpeechService mService;
+    Networking network;
+
+    protected SpeechRecognition(int aSampleRate, int aChannels, Vad aVad, Context aContext, MozillaSpeechService aService) {
+        this.mVad = aVad;
+        this.mContext = aContext;
+        this.mSampleRate = aSampleRate;
+        this.mChannels = aChannels;
+        this.mService = aService;
+    }
+
+    public void run() {
+        try {
+            Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
+
+            baos = new ByteArrayOutputStream();
+            boolean finishedvoice = false;
+            long samplesvoice = 0 ;
+            long samplessilence = 0 ;
+            boolean touchedvoice = false;
+            boolean touchedsilence = false;
+            int vad = 0;
+            long dtantes = System.currentTimeMillis();
+            long dtantesmili = 	System.currentTimeMillis();
+            boolean raisenovoice = false;
+            network = new Networking(mService);
+            network.mContext = this.mContext;
+            AudioRecord recorder = Sound.getAudioRecord(mChannels, mSampleRate);
+            EncoderInfo ef = new EncoderInfo(1, mSampleRate, 16);
+            Encoder e = Factory.getEncoder(mContext, FormatOPUS.EXT, ef, baos);
+
+            recorder.startRecording();
+            mService.notifyListeners(MozillaSpeechService.SpeechState.START_LISTEN, null);
+
+            while (!this.done && !this.cancelled) {
+                int nshorts = 0 ;
+
+                try {
+                    mBuftemp = new short[FRAME_SIZE * mChannels * 2];
+                    nshorts = recorder.read(mBuftemp, 0, mBuftemp.length);
+                    vad = mVad.feed(mBuftemp, nshorts);
+                    e.encode(mBuftemp, 0, nshorts);
+                    double[] fft =  Sound.fft(mBuftemp, 0, nshorts);
+                    double fftsum = Arrays.stream(fft).sum()/fft.length;
+                    mService.notifyListeners(MozillaSpeechService.SpeechState.MIC_ACTIVITY, fftsum);
+                }
+                catch (Exception exc) {
+                    exc.printStackTrace();
+                }
+
+                long dtdepois = System.currentTimeMillis();
+
+                if (vad == 0) {
+                    if (touchedvoice) {
+                        samplessilence += dtdepois - dtantesmili;
+                        if (samplessilence >  mMaximumSilence) touchedsilence = true;
+                    }
+                }
+                else {
+                    samplesvoice  += dtdepois - dtantesmili;
+                    if (samplesvoice >  mMinimumVoice) touchedvoice = true;
+                }
+                dtantesmili = dtdepois;
+
+                if (touchedvoice && touchedsilence)
+                    finishedvoice = true;
+
+                if (finishedvoice) {
+                    this.done = true;
+                    network.doSTT(baos);
+                }
+
+                if ((dtdepois - dtantes)/1000 > mUpperLimit ) {
+                    this.done = true;
+                    if (touchedvoice) {
+                        network.doSTT(baos);
+                    }
+                    else {
+                        raisenovoice = true;
+                    }
+                }
+
+                if (nshorts <= 0)
+                    break;
+            }
+
+            e.close();
+            mVad.stop();
+            recorder.stop();
+            recorder.release();
+
+            if (raisenovoice) mService.notifyListeners(MozillaSpeechService.SpeechState.NO_VOICE, null);
+
+            if (cancelled) {
+                cancelled = false;
+                mService.notifyListeners(MozillaSpeechService.SpeechState.CANCELED, null);
+                return;
+            }
+        }
+        catch (Exception exc)
+        {
+            String error = String.format("General audio error %s", exc.getMessage());
+            mService.notifyListeners(MozillaSpeechService.SpeechState.ERROR, error);
+            exc.printStackTrace();
+        }
+    }
+
+
+    public void cancel(){
+        cancelled = true;
+        network.cancel();
+    }
+}
diff --git a/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/Vad.java b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/Vad.java
new file mode 100644
index 0000000..658a842
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/java/com/mozilla/speechlibrary/Vad.java
@@ -0,0 +1,13 @@
+package com.mozilla.speechlibrary;
+
+public class Vad {
+
+    static {
+        System.loadLibrary("webrtc_jni");
+    }
+
+    protected native int start();
+    protected native int feed(short[] x, int n);
+    protected native int isSilence();
+    protected native int stop();
+}
diff --git a/mozillaspeechlibrary/src/main/res/values/strings.xml b/mozillaspeechlibrary/src/main/res/values/strings.xml
new file mode 100644
index 0000000..6743f56
--- /dev/null
+++ b/mozillaspeechlibrary/src/main/res/values/strings.xml
@@ -0,0 +1,3 @@
+<resources>
+    <string name="app_name">Mozilla Android SpeechLibrary</string>
+</resources>
diff --git a/mozillaspeechlibrary/src/test/java/com/mozilla/speechlibrary/ExampleUnitTest.java b/mozillaspeechlibrary/src/test/java/com/mozilla/speechlibrary/ExampleUnitTest.java
new file mode 100644
index 0000000..780aabc
--- /dev/null
+++ b/mozillaspeechlibrary/src/test/java/com/mozilla/speechlibrary/ExampleUnitTest.java
@@ -0,0 +1,17 @@
+package com.mozilla.speechlibrary;
+
+import org.junit.Test;
+
+import static org.junit.Assert.*;
+
+/**
+ * Example local unit test, which will execute on the development machine (host).
+ *
+ * @see <a href="http://d.android.com/tools/testing">Testing documentation</a>
+ */
+public class ExampleUnitTest {
+    @Test
+    public void addition_isCorrect() {
+        assertEquals(4, 2 + 2);
+    }
+}
\ No newline at end of file
diff --git a/settings.gradle b/settings.gradle
new file mode 100644
index 0000000..d99aea2
--- /dev/null
+++ b/settings.gradle
@@ -0,0 +1 @@
+include ':app', ':mozillaspeechlibrary'
\ No newline at end of file