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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /media/libsoundtouch
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'media/libsoundtouch')
-rw-r--r--media/libsoundtouch/AUTHORS4
-rw-r--r--media/libsoundtouch/LICENSE458
-rwxr-xr-xmedia/libsoundtouch/clear-lf.sh11
-rw-r--r--media/libsoundtouch/moz-libsoundtouch.patch136
-rw-r--r--media/libsoundtouch/moz.build11
-rw-r--r--media/libsoundtouch/moz.yaml96
-rw-r--r--media/libsoundtouch/src/AAFilter.cpp222
-rw-r--r--media/libsoundtouch/src/AAFilter.h93
-rw-r--r--media/libsoundtouch/src/FIFOSampleBuffer.cpp275
-rw-r--r--media/libsoundtouch/src/FIFOSampleBuffer.h180
-rw-r--r--media/libsoundtouch/src/FIFOSamplePipe.h230
-rw-r--r--media/libsoundtouch/src/FIRFilter.cpp331
-rw-r--r--media/libsoundtouch/src/FIRFilter.h140
-rw-r--r--media/libsoundtouch/src/InterpolateCubic.cpp196
-rw-r--r--media/libsoundtouch/src/InterpolateCubic.h69
-rw-r--r--media/libsoundtouch/src/InterpolateLinear.cpp296
-rw-r--r--media/libsoundtouch/src/InterpolateLinear.h98
-rw-r--r--media/libsoundtouch/src/InterpolateShannon.cpp181
-rw-r--r--media/libsoundtouch/src/InterpolateShannon.h74
-rw-r--r--media/libsoundtouch/src/RateTransposer.cpp315
-rw-r--r--media/libsoundtouch/src/RateTransposer.h164
-rw-r--r--media/libsoundtouch/src/STTypes.h195
-rw-r--r--media/libsoundtouch/src/SoundTouch.cpp538
-rw-r--r--media/libsoundtouch/src/SoundTouch.h348
-rw-r--r--media/libsoundtouch/src/SoundTouchFactory.cpp31
-rw-r--r--media/libsoundtouch/src/SoundTouchFactory.h22
-rw-r--r--media/libsoundtouch/src/TDStretch.cpp1108
-rw-r--r--media/libsoundtouch/src/TDStretch.h279
-rw-r--r--media/libsoundtouch/src/cpu_detect.h55
-rw-r--r--media/libsoundtouch/src/cpu_detect_x86.cpp130
-rw-r--r--media/libsoundtouch/src/mmx_optimized.cpp396
-rw-r--r--media/libsoundtouch/src/moz.build53
-rw-r--r--media/libsoundtouch/src/soundtouch_config.h7
-rw-r--r--media/libsoundtouch/src/soundtouch_perms.h18
-rw-r--r--media/libsoundtouch/src/sse_optimized.cpp365
-rwxr-xr-xmedia/libsoundtouch/update.sh73
36 files changed, 7198 insertions, 0 deletions
diff --git a/media/libsoundtouch/AUTHORS b/media/libsoundtouch/AUTHORS
new file mode 100644
index 0000000000..666081e195
--- /dev/null
+++ b/media/libsoundtouch/AUTHORS
@@ -0,0 +1,4 @@
+The SoundTouch Library
+Copyright © Olli Parviainen 2001-2012
+
+http://www.surina.net/soundtouch/
diff --git a/media/libsoundtouch/LICENSE b/media/libsoundtouch/LICENSE
new file mode 100644
index 0000000000..bbd24e6430
--- /dev/null
+++ b/media/libsoundtouch/LICENSE
@@ -0,0 +1,458 @@
+ GNU LESSER GENERAL PUBLIC LICENSE
+ Version 2.1, February 1999
+
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diff --git a/media/libsoundtouch/clear-lf.sh b/media/libsoundtouch/clear-lf.sh
new file mode 100755
index 0000000000..bd6295ba87
--- /dev/null
+++ b/media/libsoundtouch/clear-lf.sh
@@ -0,0 +1,11 @@
+#!/bin/bash
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+# Remove the Windows line ending characters from the files.
+for i in src/*
+do
+ cat $i | tr -d '\015' > $i.lf
+ mv $i.lf $i
+done
diff --git a/media/libsoundtouch/moz-libsoundtouch.patch b/media/libsoundtouch/moz-libsoundtouch.patch
new file mode 100644
index 0000000000..dc2ea63804
--- /dev/null
+++ b/media/libsoundtouch/moz-libsoundtouch.patch
@@ -0,0 +1,136 @@
+diff -u /src/cpu_detect_x86.cpp /src/cpu_detect_x86.cpp
+--- /src/cpu_detect_x86.cpp
++++ /src/cpu_detect_x86.cpp
+@@ -44,9 +44,8 @@
+
+
+ #if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
+-
+- #if defined(__GNUC__) && defined(__i386__)
+- // gcc
++ #if defined(__GNUC__) && defined(HAVE_CPUID_H)
++ // gcc and clang
+ #include "cpuid.h"
+ #elif defined(_M_IX86)
+ // windows non-gcc
+@@ -97,18 +96,7 @@
+
+ uint res = 0;
+
+-#if defined(__GNUC__)
+- // GCC version of cpuid. Requires GCC 4.3.0 or later for __cpuid intrinsic support.
+- uint eax, ebx, ecx, edx; // unsigned int is the standard type. uint is defined by the compiler and not guaranteed to be portable.
+-
+- // Check if no cpuid support.
+- if (!__get_cpuid (1, &eax, &ebx, &ecx, &edx)) return 0; // always disable extensions.
+-
+- if (edx & bit_MMX) res = res | SUPPORT_MMX;
+- if (edx & bit_SSE) res = res | SUPPORT_SSE;
+- if (edx & bit_SSE2) res = res | SUPPORT_SSE2;
+-
+-#else
++#if !defined(__GNUC__)
+ // Window / VS version of cpuid. Notice that Visual Studio 2005 or later required
+ // for __cpuid intrinsic support.
+ int reg[4] = {-1};
+@@ -121,7 +109,19 @@
+ if ((unsigned int)reg[3] & bit_MMX) res = res | SUPPORT_MMX;
+ if ((unsigned int)reg[3] & bit_SSE) res = res | SUPPORT_SSE;
+ if ((unsigned int)reg[3] & bit_SSE2) res = res | SUPPORT_SSE2;
++#elif defined(HAVE_CPUID_H)
++ // GCC version of cpuid. Requires GCC 4.3.0 or later for __cpuid intrinsic support.
++ uint eax, ebx, ecx, edx; // unsigned int is the standard type. uint is defined by the compiler and not guaranteed to be portable.
++
++ // Check if no cpuid support.
++ if (!__get_cpuid (1, &eax, &ebx, &ecx, &edx)) return 0; // always disable extensions.
+
++ if (edx & bit_MMX) res = res | SUPPORT_MMX;
++ if (edx & bit_SSE) res = res | SUPPORT_SSE;
++ if (edx & bit_SSE2) res = res | SUPPORT_SSE2;
++#else
++ // Compatible with GCC but no cpuid.h.
++ return 0;
+ #endif
+
+ return res & ~_dwDisabledISA;
+diff -u /src/STTypes.h /src/STTypes.h
+--- /src/STTypes.h
++++ /src/STTypes.h
+@@ -54,12 +54,17 @@
+ #define SOUNDTOUCH_ALIGN_POINTER_16(x) ( ( (ulongptr)(x) + 15 ) & ~(ulongptr)15 )
+
+
+-#if (defined(__GNUC__) && !defined(ANDROID))
+- // In GCC, include soundtouch_config.h made by config scritps.
+- // Skip this in Android compilation that uses GCC but without configure scripts.
+- #include "soundtouch_config.h"
+-#endif
++#include "soundtouch_config.h"
+
++#if defined(WIN32)
++#if defined(BUILDING_SOUNDTOUCH)
++#define SOUNDTOUCH_API __declspec(dllexport)
++#else
++#define SOUNDTOUCH_API __declspec(dllimport)
++#endif
++#else
++#define SOUNDTOUCH_API
++#endif
+
+ namespace soundtouch
+ {
+diff -u /src/SoundTouch.h /src/SoundTouch.h
+--- /src/SoundTouch.h
++++ /src/SoundTouch.h
+@@ -141,7 +141,7 @@
+ /// tempo/pitch/rate/samplerate settings.
+ #define SETTING_NOMINAL_OUTPUT_SEQUENCE 7
+
+-class SoundTouch : public FIFOProcessor
++class SOUNDTOUCH_API SoundTouch : public FIFOProcessor
+ {
+ private:
+ /// Rate transposer class instance
+diff -u /src/FIRFilter.cpp /src/FIRFilter.cpp
+--- /src/FIRFilter.cpp
++++ /src/FIRFilter.cpp
+@@ -291,9 +296,11 @@
+
+ FIRFilter * FIRFilter::newInstance()
+ {
++#if defined(SOUNDTOUCH_ALLOW_MMX) || defined(SOUNDTOUCH_ALLOW_SSE)
+ uint uExtensions;
+
+ uExtensions = detectCPUextensions();
++#endif
+
+ // Check if MMX/SSE instruction set extensions supported by CPU
+
+diff -u /src/TDStretch.cpp /src/TDStretch.cpp
+--- /src/TDStretch.cpp
++++ /src/TDStretch.cpp
+@@ -624,9 +624,11 @@
+
+ TDStretch * TDStretch::newInstance()
+ {
++#if defined(SOUNDTOUCH_ALLOW_MMX) || defined(SOUNDTOUCH_ALLOW_SSE)
+ uint uExtensions;
+
+ uExtensions = detectCPUextensions();
++#endif
+
+ // Check if MMX/SSE instruction set extensions supported by CPU
+
+diff -u /src/AAFilter.cpp /src/AAFilter.cpp
+--- /src/AAFilter.cpp
++++ /src/AAFilter.cpp
+@@ -42,7 +42,7 @@
+
+ using namespace soundtouch;
+
+-#define PI 3.14159265358979323846
++#define PI M_PI
+ #define TWOPI (2 * PI)
+
+ // define this to save AA filter coefficients to a file
+
diff --git a/media/libsoundtouch/moz.build b/media/libsoundtouch/moz.build
new file mode 100644
index 0000000000..b948b4ccfc
--- /dev/null
+++ b/media/libsoundtouch/moz.build
@@ -0,0 +1,11 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+with Files("**"):
+ BUG_COMPONENT = ("Core", "Audio/Video: Playback")
+
+DIRS += ['src']
+
diff --git a/media/libsoundtouch/moz.yaml b/media/libsoundtouch/moz.yaml
new file mode 100644
index 0000000000..f3d0da4516
--- /dev/null
+++ b/media/libsoundtouch/moz.yaml
@@ -0,0 +1,96 @@
+schema: 1
+
+bugzilla:
+ product: "Core"
+ component: "Audio/Video"
+
+# The whole library is not used, only the relevant files are imported in the
+# tree, using the script `update.sh`. Some changes have been made to the files,
+# using the patch `moz-libsoundtouch.patch`.
+# We also use a custom soundtouch_config.h.
+
+origin:
+ name: "SoundTouch"
+ description: "Audio processing library"
+
+ url: "http://www.surina.net/soundtouch"
+ license: "LGPL-2.1"
+
+ release: 17a63e99d58053b0fbde8b69d7cc76a38119ce70 (2021-03-03T18:11:45+02:00).
+ revision: 17a63e99d58053b0fbde8b69d7cc76a38119ce70
+
+vendoring:
+ url: https://codeberg.org/soundtouch/soundtouch
+ source-hosting: codeberg
+ tracking: commit
+
+ exclude:
+ - bin
+ - bootstrap
+ - config
+ - configure.ac
+ - createsrcpack
+ - Makefile.am
+ - make-win.bat
+ - README.html
+ - readme.md
+ - soundtouch.m4
+ - soundtouch.pc.in
+ - ".*"
+ - "source/Android-lib"
+ - "source/*/*/*"
+ - "source/*/*"
+ - "source/*/.*"
+ - "source/*"
+ - "include/*"
+
+ include:
+ - COPYING.TXT
+ - include/SoundTouch.h
+ - include/FIFOSampleBuffer.h
+ - include/FIFOSamplePipe.h
+ - include/STTypes.h
+ - source/SoundTouch/AAFilter.cpp
+ - source/SoundTouch/AAFilter.h
+ - source/SoundTouch/cpu_detect.h
+ - source/SoundTouch/cpu_detect_x86.cpp
+ - source/SoundTouch/FIFOSampleBuffer.cpp
+ - source/SoundTouch/FIRFilter.cpp
+ - source/SoundTouch/FIRFilter.h
+ - source/SoundTouch/InterpolateLinear.cpp
+ - source/SoundTouch/InterpolateLinear.h
+ - source/SoundTouch/InterpolateCubic.cpp
+ - source/SoundTouch/InterpolateCubic.h
+ - source/SoundTouch/InterpolateShannon.cpp
+ - source/SoundTouch/InterpolateShannon.h
+ - source/SoundTouch/mmx_optimized.cpp
+ - source/SoundTouch/RateTransposer.cpp
+ - source/SoundTouch/RateTransposer.h
+ - source/SoundTouch/SoundTouch.cpp
+ - source/SoundTouch/sse_optimized.cpp
+ - source/SoundTouch/TDStretch.cpp
+ - source/SoundTouch/TDStretch.h
+
+ keep:
+ - AUTHORS
+ - clear-lf.sh
+ - src/SoundTouchFactory.*
+ - src/soundtouch_config.h
+ - src/soundtouch_perms.h
+
+ patches:
+ - moz-libsoundtouch.patch
+
+ update-actions:
+ - action: move-file
+ from: '{vendor_dir}/COPYING.TXT'
+ to: '{vendor_dir}/LICENSE'
+ - action: move-dir
+ from: '{vendor_dir}/source/SoundTouch'
+ to: '{vendor_dir}/src'
+ - action: move-dir
+ from: '{vendor_dir}/include'
+ to: '{vendor_dir}/src'
+ - action: run-script
+ script: '{yaml_dir}/clear-lf.sh'
+ cwd: '{yaml_dir}'
diff --git a/media/libsoundtouch/src/AAFilter.cpp b/media/libsoundtouch/src/AAFilter.cpp
new file mode 100644
index 0000000000..3fd4e9fe6d
--- /dev/null
+++ b/media/libsoundtouch/src/AAFilter.cpp
@@ -0,0 +1,222 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
+/// MMX optimization.
+///
+/// Anti-alias filter is used to prevent folding of high frequencies when
+/// transposing the sample rate with interpolation.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include "AAFilter.h"
+#include "FIRFilter.h"
+
+using namespace soundtouch;
+
+#define PI M_PI
+#define TWOPI (2 * PI)
+
+// define this to save AA filter coefficients to a file
+// #define _DEBUG_SAVE_AAFILTER_COEFFICIENTS 1
+
+#ifdef _DEBUG_SAVE_AAFILTER_COEFFICIENTS
+ #include <stdio.h>
+
+ static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len)
+ {
+ FILE *fptr = fopen("aa_filter_coeffs.txt", "wt");
+ if (fptr == NULL) return;
+
+ for (int i = 0; i < len; i ++)
+ {
+ double temp = coeffs[i];
+ fprintf(fptr, "%lf\n", temp);
+ }
+ fclose(fptr);
+ }
+
+#else
+ #define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
+#endif
+
+/*****************************************************************************
+ *
+ * Implementation of the class 'AAFilter'
+ *
+ *****************************************************************************/
+
+AAFilter::AAFilter(uint len)
+{
+ pFIR = FIRFilter::newInstance();
+ cutoffFreq = 0.5;
+ setLength(len);
+}
+
+
+AAFilter::~AAFilter()
+{
+ delete pFIR;
+}
+
+
+// Sets new anti-alias filter cut-off edge frequency, scaled to
+// sampling frequency (nyquist frequency = 0.5).
+// The filter will cut frequencies higher than the given frequency.
+void AAFilter::setCutoffFreq(double newCutoffFreq)
+{
+ cutoffFreq = newCutoffFreq;
+ calculateCoeffs();
+}
+
+
+// Sets number of FIR filter taps
+void AAFilter::setLength(uint newLength)
+{
+ length = newLength;
+ calculateCoeffs();
+}
+
+
+// Calculates coefficients for a low-pass FIR filter using Hamming window
+void AAFilter::calculateCoeffs()
+{
+ uint i;
+ double cntTemp, temp, tempCoeff,h, w;
+ double wc;
+ double scaleCoeff, sum;
+ double *work;
+ SAMPLETYPE *coeffs;
+
+ assert(length >= 2);
+ assert(length % 4 == 0);
+ assert(cutoffFreq >= 0);
+ assert(cutoffFreq <= 0.5);
+
+ work = new double[length];
+ coeffs = new SAMPLETYPE[length];
+
+ wc = 2.0 * PI * cutoffFreq;
+ tempCoeff = TWOPI / (double)length;
+
+ sum = 0;
+ for (i = 0; i < length; i ++)
+ {
+ cntTemp = (double)i - (double)(length / 2);
+
+ temp = cntTemp * wc;
+ if (temp != 0)
+ {
+ h = sin(temp) / temp; // sinc function
+ }
+ else
+ {
+ h = 1.0;
+ }
+ w = 0.54 + 0.46 * cos(tempCoeff * cntTemp); // hamming window
+
+ temp = w * h;
+ work[i] = temp;
+
+ // calc net sum of coefficients
+ sum += temp;
+ }
+
+ // ensure the sum of coefficients is larger than zero
+ assert(sum > 0);
+
+ // ensure we've really designed a lowpass filter...
+ assert(work[length/2] > 0);
+ assert(work[length/2 + 1] > -1e-6);
+ assert(work[length/2 - 1] > -1e-6);
+
+ // Calculate a scaling coefficient in such a way that the result can be
+ // divided by 16384
+ scaleCoeff = 16384.0f / sum;
+
+ for (i = 0; i < length; i ++)
+ {
+ temp = work[i] * scaleCoeff;
+ // scale & round to nearest integer
+ temp += (temp >= 0) ? 0.5 : -0.5;
+ // ensure no overfloods
+ assert(temp >= -32768 && temp <= 32767);
+ coeffs[i] = (SAMPLETYPE)temp;
+ }
+
+ // Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
+ pFIR->setCoefficients(coeffs, length, 14);
+
+ _DEBUG_SAVE_AAFIR_COEFFS(coeffs, length);
+
+ delete[] work;
+ delete[] coeffs;
+}
+
+
+// Applies the filter to the given sequence of samples.
+// Note : The amount of outputted samples is by value of 'filter length'
+// smaller than the amount of input samples.
+uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
+{
+ return pFIR->evaluate(dest, src, numSamples, numChannels);
+}
+
+
+/// Applies the filter to the given src & dest pipes, so that processed amount of
+/// samples get removed from src, and produced amount added to dest
+/// Note : The amount of outputted samples is by value of 'filter length'
+/// smaller than the amount of input samples.
+uint AAFilter::evaluate(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) const
+{
+ SAMPLETYPE *pdest;
+ const SAMPLETYPE *psrc;
+ uint numSrcSamples;
+ uint result;
+ int numChannels = src.getChannels();
+
+ assert(numChannels == dest.getChannels());
+
+ numSrcSamples = src.numSamples();
+ psrc = src.ptrBegin();
+ pdest = dest.ptrEnd(numSrcSamples);
+ result = pFIR->evaluate(pdest, psrc, numSrcSamples, numChannels);
+ src.receiveSamples(result);
+ dest.putSamples(result);
+
+ return result;
+}
+
+
+uint AAFilter::getLength() const
+{
+ return pFIR->getLength();
+}
diff --git a/media/libsoundtouch/src/AAFilter.h b/media/libsoundtouch/src/AAFilter.h
new file mode 100644
index 0000000000..81d836b750
--- /dev/null
+++ b/media/libsoundtouch/src/AAFilter.h
@@ -0,0 +1,93 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
+/// while maintaining the original pitch by using a time domain WSOLA-like method
+/// with several performance-increasing tweaks.
+///
+/// Anti-alias filter is used to prevent folding of high frequencies when
+/// transposing the sample rate with interpolation.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef AAFilter_H
+#define AAFilter_H
+
+#include "STTypes.h"
+#include "FIFOSampleBuffer.h"
+
+namespace soundtouch
+{
+
+class AAFilter
+{
+protected:
+ class FIRFilter *pFIR;
+
+ /// Low-pass filter cut-off frequency, negative = invalid
+ double cutoffFreq;
+
+ /// num of filter taps
+ uint length;
+
+ /// Calculate the FIR coefficients realizing the given cutoff-frequency
+ void calculateCoeffs();
+public:
+ AAFilter(uint length);
+
+ ~AAFilter();
+
+ /// Sets new anti-alias filter cut-off edge frequency, scaled to sampling
+ /// frequency (nyquist frequency = 0.5). The filter will cut off the
+ /// frequencies than that.
+ void setCutoffFreq(double newCutoffFreq);
+
+ /// Sets number of FIR filter taps, i.e. ~filter complexity
+ void setLength(uint newLength);
+
+ uint getLength() const;
+
+ /// Applies the filter to the given sequence of samples.
+ /// Note : The amount of outputted samples is by value of 'filter length'
+ /// smaller than the amount of input samples.
+ uint evaluate(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples,
+ uint numChannels) const;
+
+ /// Applies the filter to the given src & dest pipes, so that processed amount of
+ /// samples get removed from src, and produced amount added to dest
+ /// Note : The amount of outputted samples is by value of 'filter length'
+ /// smaller than the amount of input samples.
+ uint evaluate(FIFOSampleBuffer &dest,
+ FIFOSampleBuffer &src) const;
+
+};
+
+}
+
+#endif
diff --git a/media/libsoundtouch/src/FIFOSampleBuffer.cpp b/media/libsoundtouch/src/FIFOSampleBuffer.cpp
new file mode 100644
index 0000000000..ad36875466
--- /dev/null
+++ b/media/libsoundtouch/src/FIFOSampleBuffer.cpp
@@ -0,0 +1,275 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// A buffer class for temporarily storaging sound samples, operates as a
+/// first-in-first-out pipe.
+///
+/// Samples are added to the end of the sample buffer with the 'putSamples'
+/// function, and are received from the beginning of the buffer by calling
+/// the 'receiveSamples' function. The class automatically removes the
+/// outputted samples from the buffer, as well as grows the buffer size
+/// whenever necessary.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <stdlib.h>
+#include <memory.h>
+#include <string.h>
+#include <assert.h>
+
+#include "FIFOSampleBuffer.h"
+
+using namespace soundtouch;
+
+// Constructor
+FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
+{
+ assert(numChannels > 0);
+ sizeInBytes = 0; // reasonable initial value
+ buffer = NULL;
+ bufferUnaligned = NULL;
+ samplesInBuffer = 0;
+ bufferPos = 0;
+ channels = (uint)numChannels;
+ ensureCapacity(32); // allocate initial capacity
+}
+
+
+// destructor
+FIFOSampleBuffer::~FIFOSampleBuffer()
+{
+ delete[] bufferUnaligned;
+ bufferUnaligned = NULL;
+ buffer = NULL;
+}
+
+
+// Sets number of channels, 1 = mono, 2 = stereo
+void FIFOSampleBuffer::setChannels(int numChannels)
+{
+ uint usedBytes;
+
+ if (!verifyNumberOfChannels(numChannels)) return;
+
+ usedBytes = channels * samplesInBuffer;
+ channels = (uint)numChannels;
+ samplesInBuffer = usedBytes / channels;
+}
+
+
+// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and
+// zeroes this pointer by copying samples from the 'bufferPos' pointer
+// location on to the beginning of the buffer.
+void FIFOSampleBuffer::rewind()
+{
+ if (buffer && bufferPos)
+ {
+ memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer);
+ bufferPos = 0;
+ }
+}
+
+
+// Adds 'numSamples' pcs of samples from the 'samples' memory position to
+// the sample buffer.
+void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint nSamples)
+{
+ memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels);
+ samplesInBuffer += nSamples;
+}
+
+
+// Increases the number of samples in the buffer without copying any actual
+// samples.
+//
+// This function is used to update the number of samples in the sample buffer
+// when accessing the buffer directly with 'ptrEnd' function. Please be
+// careful though!
+void FIFOSampleBuffer::putSamples(uint nSamples)
+{
+ uint req;
+
+ req = samplesInBuffer + nSamples;
+ ensureCapacity(req);
+ samplesInBuffer += nSamples;
+}
+
+
+// Returns a pointer to the end of the used part of the sample buffer (i.e.
+// where the new samples are to be inserted). This function may be used for
+// inserting new samples into the sample buffer directly. Please be careful!
+//
+// Parameter 'slackCapacity' tells the function how much free capacity (in
+// terms of samples) there _at least_ should be, in order to the caller to
+// successfully insert all the required samples to the buffer. When necessary,
+// the function grows the buffer size to comply with this requirement.
+//
+// When using this function as means for inserting new samples, also remember
+// to increase the sample count afterwards, by calling the
+// 'putSamples(numSamples)' function.
+SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity)
+{
+ ensureCapacity(samplesInBuffer + slackCapacity);
+ return buffer + samplesInBuffer * channels;
+}
+
+
+// Returns a pointer to the beginning of the currently non-outputted samples.
+// This function is provided for accessing the output samples directly.
+// Please be careful!
+//
+// When using this function to output samples, also remember to 'remove' the
+// outputted samples from the buffer by calling the
+// 'receiveSamples(numSamples)' function
+SAMPLETYPE *FIFOSampleBuffer::ptrBegin()
+{
+ assert(buffer);
+ return buffer + bufferPos * channels;
+}
+
+
+// Ensures that the buffer has enough capacity, i.e. space for _at least_
+// 'capacityRequirement' number of samples. The buffer is grown in steps of
+// 4 kilobytes to eliminate the need for frequently growing up the buffer,
+// as well as to round the buffer size up to the virtual memory page size.
+void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
+{
+ SAMPLETYPE *tempUnaligned, *temp;
+
+ if (capacityRequirement > getCapacity())
+ {
+ // enlarge the buffer in 4kbyte steps (round up to next 4k boundary)
+ sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;
+ assert(sizeInBytes % 2 == 0);
+ tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
+ if (tempUnaligned == NULL)
+ {
+ ST_THROW_RT_ERROR("Couldn't allocate memory!\n");
+ }
+ // Align the buffer to begin at 16byte cache line boundary for optimal performance
+ temp = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(tempUnaligned);
+ if (samplesInBuffer)
+ {
+ memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
+ }
+ delete[] bufferUnaligned;
+ buffer = temp;
+ bufferUnaligned = tempUnaligned;
+ bufferPos = 0;
+ }
+ else
+ {
+ // simply rewind the buffer (if necessary)
+ rewind();
+ }
+}
+
+
+// Returns the current buffer capacity in terms of samples
+uint FIFOSampleBuffer::getCapacity() const
+{
+ return sizeInBytes / (channels * sizeof(SAMPLETYPE));
+}
+
+
+// Returns the number of samples currently in the buffer
+uint FIFOSampleBuffer::numSamples() const
+{
+ return samplesInBuffer;
+}
+
+
+// Output samples from beginning of the sample buffer. Copies demanded number
+// of samples to output and removes them from the sample buffer. If there
+// are less than 'numsample' samples in the buffer, returns all available.
+//
+// Returns number of samples copied.
+uint FIFOSampleBuffer::receiveSamples(SAMPLETYPE *output, uint maxSamples)
+{
+ uint num;
+
+ num = (maxSamples > samplesInBuffer) ? samplesInBuffer : maxSamples;
+
+ memcpy(output, ptrBegin(), channels * sizeof(SAMPLETYPE) * num);
+ return receiveSamples(num);
+}
+
+
+// Removes samples from the beginning of the sample buffer without copying them
+// anywhere. Used to reduce the number of samples in the buffer, when accessing
+// the sample buffer with the 'ptrBegin' function.
+uint FIFOSampleBuffer::receiveSamples(uint maxSamples)
+{
+ if (maxSamples >= samplesInBuffer)
+ {
+ uint temp;
+
+ temp = samplesInBuffer;
+ samplesInBuffer = 0;
+ return temp;
+ }
+
+ samplesInBuffer -= maxSamples;
+ bufferPos += maxSamples;
+
+ return maxSamples;
+}
+
+
+// Returns nonzero if the sample buffer is empty
+int FIFOSampleBuffer::isEmpty() const
+{
+ return (samplesInBuffer == 0) ? 1 : 0;
+}
+
+
+// Clears the sample buffer
+void FIFOSampleBuffer::clear()
+{
+ samplesInBuffer = 0;
+ bufferPos = 0;
+}
+
+
+/// allow trimming (downwards) amount of samples in pipeline.
+/// Returns adjusted amount of samples
+uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
+{
+ if (numSamples < samplesInBuffer)
+ {
+ samplesInBuffer = numSamples;
+ }
+ return samplesInBuffer;
+}
+
+
+/// Add silence to end of buffer
+void FIFOSampleBuffer::addSilent(uint nSamples)
+{
+ memset(ptrEnd(nSamples), 0, sizeof(SAMPLETYPE) * nSamples * channels);
+ samplesInBuffer += nSamples;
+}
diff --git a/media/libsoundtouch/src/FIFOSampleBuffer.h b/media/libsoundtouch/src/FIFOSampleBuffer.h
new file mode 100644
index 0000000000..537a7b8722
--- /dev/null
+++ b/media/libsoundtouch/src/FIFOSampleBuffer.h
@@ -0,0 +1,180 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// A buffer class for temporarily storaging sound samples, operates as a
+/// first-in-first-out pipe.
+///
+/// Samples are added to the end of the sample buffer with the 'putSamples'
+/// function, and are received from the beginning of the buffer by calling
+/// the 'receiveSamples' function. The class automatically removes the
+/// output samples from the buffer as well as grows the storage size
+/// whenever necessary.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef FIFOSampleBuffer_H
+#define FIFOSampleBuffer_H
+
+#include "FIFOSamplePipe.h"
+
+namespace soundtouch
+{
+
+/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes
+/// care of storage size adjustment and data moving during input/output operations.
+///
+/// Notice that in case of stereo audio, one sample is considered to consist of
+/// both channel data.
+class FIFOSampleBuffer : public FIFOSamplePipe
+{
+private:
+ /// Sample buffer.
+ SAMPLETYPE *buffer;
+
+ // Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first
+ // 16-byte aligned location of this buffer
+ SAMPLETYPE *bufferUnaligned;
+
+ /// Sample buffer size in bytes
+ uint sizeInBytes;
+
+ /// How many samples are currently in buffer.
+ uint samplesInBuffer;
+
+ /// Channels, 1=mono, 2=stereo.
+ uint channels;
+
+ /// Current position pointer to the buffer. This pointer is increased when samples are
+ /// removed from the pipe so that it's necessary to actually rewind buffer (move data)
+ /// only new data when is put to the pipe.
+ uint bufferPos;
+
+ /// Rewind the buffer by moving data from position pointed by 'bufferPos' to real
+ /// beginning of the buffer.
+ void rewind();
+
+ /// Ensures that the buffer has capacity for at least this many samples.
+ void ensureCapacity(uint capacityRequirement);
+
+ /// Returns current capacity.
+ uint getCapacity() const;
+
+public:
+
+ /// Constructor
+ FIFOSampleBuffer(int numChannels = 2 ///< Number of channels, 1=mono, 2=stereo.
+ ///< Default is stereo.
+ );
+
+ /// destructor
+ ~FIFOSampleBuffer();
+
+ /// Returns a pointer to the beginning of the output samples.
+ /// This function is provided for accessing the output samples directly.
+ /// Please be careful for not to corrupt the book-keeping!
+ ///
+ /// When using this function to output samples, also remember to 'remove' the
+ /// output samples from the buffer by calling the
+ /// 'receiveSamples(numSamples)' function
+ virtual SAMPLETYPE *ptrBegin();
+
+ /// Returns a pointer to the end of the used part of the sample buffer (i.e.
+ /// where the new samples are to be inserted). This function may be used for
+ /// inserting new samples into the sample buffer directly. Please be careful
+ /// not corrupt the book-keeping!
+ ///
+ /// When using this function as means for inserting new samples, also remember
+ /// to increase the sample count afterwards, by calling the
+ /// 'putSamples(numSamples)' function.
+ SAMPLETYPE *ptrEnd(
+ uint slackCapacity ///< How much free capacity (in samples) there _at least_
+ ///< should be so that the caller can successfully insert the
+ ///< desired samples to the buffer. If necessary, the function
+ ///< grows the buffer size to comply with this requirement.
+ );
+
+ /// Adds 'numSamples' pcs of samples from the 'samples' memory position to
+ /// the sample buffer.
+ virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
+ uint numSamples ///< Number of samples to insert.
+ );
+
+ /// Adjusts the book-keeping to increase number of samples in the buffer without
+ /// copying any actual samples.
+ ///
+ /// This function is used to update the number of samples in the sample buffer
+ /// when accessing the buffer directly with 'ptrEnd' function. Please be
+ /// careful though!
+ virtual void putSamples(uint numSamples ///< Number of samples been inserted.
+ );
+
+ /// Output samples from beginning of the sample buffer. Copies requested samples to
+ /// output buffer and removes them from the sample buffer. If there are less than
+ /// 'numsample' samples in the buffer, returns all that available.
+ ///
+ /// \return Number of samples returned.
+ virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
+ uint maxSamples ///< How many samples to receive at max.
+ );
+
+ /// Adjusts book-keeping so that given number of samples are removed from beginning of the
+ /// sample buffer without copying them anywhere.
+ ///
+ /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+ /// with 'ptrBegin' function.
+ virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
+ );
+
+ /// Returns number of samples currently available.
+ virtual uint numSamples() const;
+
+ /// Sets number of channels, 1 = mono, 2 = stereo.
+ void setChannels(int numChannels);
+
+ /// Get number of channels
+ int getChannels()
+ {
+ return channels;
+ }
+
+ /// Returns nonzero if there aren't any samples available for outputting.
+ virtual int isEmpty() const;
+
+ /// Clears all the samples.
+ virtual void clear();
+
+ /// allow trimming (downwards) amount of samples in pipeline.
+ /// Returns adjusted amount of samples
+ uint adjustAmountOfSamples(uint numSamples);
+
+ /// Add silence to end of buffer
+ void addSilent(uint nSamples);
+};
+
+}
+
+#endif
diff --git a/media/libsoundtouch/src/FIFOSamplePipe.h b/media/libsoundtouch/src/FIFOSamplePipe.h
new file mode 100644
index 0000000000..3def42d1ab
--- /dev/null
+++ b/media/libsoundtouch/src/FIFOSamplePipe.h
@@ -0,0 +1,230 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound
+/// samples by operating like a first-in-first-out pipe: New samples are fed
+/// into one end of the pipe with the 'putSamples' function, and the processed
+/// samples are received from the other end with the 'receiveSamples' function.
+///
+/// 'FIFOProcessor' : A base class for classes the do signal processing with
+/// the samples while operating like a first-in-first-out pipe. When samples
+/// are input with the 'putSamples' function, the class processes them
+/// and moves the processed samples to the given 'output' pipe object, which
+/// may be either another processing stage, or a fifo sample buffer object.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef FIFOSamplePipe_H
+#define FIFOSamplePipe_H
+
+#include <assert.h>
+#include <stdlib.h>
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// Abstract base class for FIFO (first-in-first-out) sample processing classes.
+class FIFOSamplePipe
+{
+protected:
+
+ bool verifyNumberOfChannels(int nChannels) const
+ {
+ if ((nChannels > 0) && (nChannels <= SOUNDTOUCH_MAX_CHANNELS))
+ {
+ return true;
+ }
+ ST_THROW_RT_ERROR("Error: Illegal number of channels");
+ return false;
+ }
+
+public:
+ // virtual default destructor
+ virtual ~FIFOSamplePipe() {}
+
+
+ /// Returns a pointer to the beginning of the output samples.
+ /// This function is provided for accessing the output samples directly.
+ /// Please be careful for not to corrupt the book-keeping!
+ ///
+ /// When using this function to output samples, also remember to 'remove' the
+ /// output samples from the buffer by calling the
+ /// 'receiveSamples(numSamples)' function
+ virtual SAMPLETYPE *ptrBegin() = 0;
+
+ /// Adds 'numSamples' pcs of samples from the 'samples' memory position to
+ /// the sample buffer.
+ virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
+ uint numSamples ///< Number of samples to insert.
+ ) = 0;
+
+
+ // Moves samples from the 'other' pipe instance to this instance.
+ void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data.
+ )
+ {
+ int oNumSamples = other.numSamples();
+
+ putSamples(other.ptrBegin(), oNumSamples);
+ other.receiveSamples(oNumSamples);
+ };
+
+ /// Output samples from beginning of the sample buffer. Copies requested samples to
+ /// output buffer and removes them from the sample buffer. If there are less than
+ /// 'numsample' samples in the buffer, returns all that available.
+ ///
+ /// \return Number of samples returned.
+ virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
+ uint maxSamples ///< How many samples to receive at max.
+ ) = 0;
+
+ /// Adjusts book-keeping so that given number of samples are removed from beginning of the
+ /// sample buffer without copying them anywhere.
+ ///
+ /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+ /// with 'ptrBegin' function.
+ virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
+ ) = 0;
+
+ /// Returns number of samples currently available.
+ virtual uint numSamples() const = 0;
+
+ // Returns nonzero if there aren't any samples available for outputting.
+ virtual int isEmpty() const = 0;
+
+ /// Clears all the samples.
+ virtual void clear() = 0;
+
+ /// allow trimming (downwards) amount of samples in pipeline.
+ /// Returns adjusted amount of samples
+ virtual uint adjustAmountOfSamples(uint numSamples) = 0;
+
+};
+
+
+/// Base-class for sound processing routines working in FIFO principle. With this base
+/// class it's easy to implement sound processing stages that can be chained together,
+/// so that samples that are fed into beginning of the pipe automatically go through
+/// all the processing stages.
+///
+/// When samples are input to this class, they're first processed and then put to
+/// the FIFO pipe that's defined as output of this class. This output pipe can be
+/// either other processing stage or a FIFO sample buffer.
+class FIFOProcessor :public FIFOSamplePipe
+{
+protected:
+ /// Internal pipe where processed samples are put.
+ FIFOSamplePipe *output;
+
+ /// Sets output pipe.
+ void setOutPipe(FIFOSamplePipe *pOutput)
+ {
+ assert(output == NULL);
+ assert(pOutput != NULL);
+ output = pOutput;
+ }
+
+ /// Constructor. Doesn't define output pipe; it has to be set be
+ /// 'setOutPipe' function.
+ FIFOProcessor()
+ {
+ output = NULL;
+ }
+
+ /// Constructor. Configures output pipe.
+ FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
+ )
+ {
+ output = pOutput;
+ }
+
+ /// Destructor.
+ virtual ~FIFOProcessor()
+ {
+ }
+
+ /// Returns a pointer to the beginning of the output samples.
+ /// This function is provided for accessing the output samples directly.
+ /// Please be careful for not to corrupt the book-keeping!
+ ///
+ /// When using this function to output samples, also remember to 'remove' the
+ /// output samples from the buffer by calling the
+ /// 'receiveSamples(numSamples)' function
+ virtual SAMPLETYPE *ptrBegin()
+ {
+ return output->ptrBegin();
+ }
+
+public:
+
+ /// Output samples from beginning of the sample buffer. Copies requested samples to
+ /// output buffer and removes them from the sample buffer. If there are less than
+ /// 'numsample' samples in the buffer, returns all that available.
+ ///
+ /// \return Number of samples returned.
+ virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
+ uint maxSamples ///< How many samples to receive at max.
+ )
+ {
+ return output->receiveSamples(outBuffer, maxSamples);
+ }
+
+ /// Adjusts book-keeping so that given number of samples are removed from beginning of the
+ /// sample buffer without copying them anywhere.
+ ///
+ /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+ /// with 'ptrBegin' function.
+ virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
+ )
+ {
+ return output->receiveSamples(maxSamples);
+ }
+
+ /// Returns number of samples currently available.
+ virtual uint numSamples() const
+ {
+ return output->numSamples();
+ }
+
+ /// Returns nonzero if there aren't any samples available for outputting.
+ virtual int isEmpty() const
+ {
+ return output->isEmpty();
+ }
+
+ /// allow trimming (downwards) amount of samples in pipeline.
+ /// Returns adjusted amount of samples
+ virtual uint adjustAmountOfSamples(uint numSamples)
+ {
+ return output->adjustAmountOfSamples(numSamples);
+ }
+};
+
+}
+
+#endif
diff --git a/media/libsoundtouch/src/FIRFilter.cpp b/media/libsoundtouch/src/FIRFilter.cpp
new file mode 100644
index 0000000000..201395bd51
--- /dev/null
+++ b/media/libsoundtouch/src/FIRFilter.cpp
@@ -0,0 +1,331 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// General FIR digital filter routines with MMX optimization.
+///
+/// Notes : MMX optimized functions reside in a separate, platform-specific file,
+/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
+///
+/// This source file contains OpenMP optimizations that allow speeding up the
+/// corss-correlation algorithm by executing it in several threads / CPU cores
+/// in parallel. See the following article link for more detailed discussion
+/// about SoundTouch OpenMP optimizations:
+/// http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+#include "FIRFilter.h"
+#include "cpu_detect.h"
+
+using namespace soundtouch;
+
+/*****************************************************************************
+ *
+ * Implementation of the class 'FIRFilter'
+ *
+ *****************************************************************************/
+
+FIRFilter::FIRFilter()
+{
+ resultDivFactor = 0;
+ resultDivider = 0;
+ length = 0;
+ lengthDiv8 = 0;
+ filterCoeffs = NULL;
+ filterCoeffsStereo = NULL;
+}
+
+
+FIRFilter::~FIRFilter()
+{
+ delete[] filterCoeffs;
+ delete[] filterCoeffsStereo;
+}
+
+
+// Usual C-version of the filter routine for stereo sound
+uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
+{
+ int j, end;
+#ifdef SOUNDTOUCH_FLOAT_SAMPLES
+ // when using floating point samples, use a scaler instead of a divider
+ // because division is much slower operation than multiplying.
+ double dScaler = 1.0 / (double)resultDivider;
+#endif
+ // hint compiler autovectorization that loop length is divisible by 8
+ int ilength = length & -8;
+
+ assert((length != 0) && (length == ilength) && (src != NULL) && (dest != NULL) && (filterCoeffs != NULL));
+
+ end = 2 * (numSamples - ilength);
+
+ #pragma omp parallel for
+ for (j = 0; j < end; j += 2)
+ {
+ const SAMPLETYPE *ptr;
+ LONG_SAMPLETYPE suml, sumr;
+
+ suml = sumr = 0;
+ ptr = src + j;
+
+ for (int i = 0; i < ilength; i ++)
+ {
+ suml += ptr[2 * i] * filterCoeffsStereo[2 * i];
+ sumr += ptr[2 * i + 1] * filterCoeffsStereo[2 * i + 1];
+ }
+
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+ suml >>= resultDivFactor;
+ sumr >>= resultDivFactor;
+ // saturate to 16 bit integer limits
+ suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
+ // saturate to 16 bit integer limits
+ sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+ dest[j] = (SAMPLETYPE)suml;
+ dest[j + 1] = (SAMPLETYPE)sumr;
+ }
+ return numSamples - ilength;
+}
+
+
+// Usual C-version of the filter routine for mono sound
+uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
+{
+ int j, end;
+#ifdef SOUNDTOUCH_FLOAT_SAMPLES
+ // when using floating point samples, use a scaler instead of a divider
+ // because division is much slower operation than multiplying.
+ double dScaler = 1.0 / (double)resultDivider;
+#endif
+
+ // hint compiler autovectorization that loop length is divisible by 8
+ int ilength = length & -8;
+
+ assert(ilength != 0);
+
+ end = numSamples - ilength;
+ #pragma omp parallel for
+ for (j = 0; j < end; j ++)
+ {
+ const SAMPLETYPE *pSrc = src + j;
+ LONG_SAMPLETYPE sum;
+ int i;
+
+ sum = 0;
+ for (i = 0; i < ilength; i ++)
+ {
+ sum += pSrc[i] * filterCoeffs[i];
+ }
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+ sum >>= resultDivFactor;
+ // saturate to 16 bit integer limits
+ sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+ dest[j] = (SAMPLETYPE)sum;
+ }
+ return end;
+}
+
+
+uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
+{
+ int j, end;
+
+#ifdef SOUNDTOUCH_FLOAT_SAMPLES
+ // when using floating point samples, use a scaler instead of a divider
+ // because division is much slower operation than multiplying.
+ double dScaler = 1.0 / (double)resultDivider;
+#endif
+
+ assert(length != 0);
+ assert(src != NULL);
+ assert(dest != NULL);
+ assert(filterCoeffs != NULL);
+ assert(numChannels < 16);
+
+ // hint compiler autovectorization that loop length is divisible by 8
+ int ilength = length & -8;
+
+ end = numChannels * (numSamples - ilength);
+
+ #pragma omp parallel for
+ for (j = 0; j < end; j += numChannels)
+ {
+ const SAMPLETYPE *ptr;
+ LONG_SAMPLETYPE sums[16];
+ uint c;
+ int i;
+
+ for (c = 0; c < numChannels; c ++)
+ {
+ sums[c] = 0;
+ }
+
+ ptr = src + j;
+
+ for (i = 0; i < ilength; i ++)
+ {
+ SAMPLETYPE coef=filterCoeffs[i];
+ for (c = 0; c < numChannels; c ++)
+ {
+ sums[c] += ptr[0] * coef;
+ ptr ++;
+ }
+ }
+
+ for (c = 0; c < numChannels; c ++)
+ {
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+ sums[c] >>= resultDivFactor;
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+ dest[j+c] = (SAMPLETYPE)sums[c];
+ }
+ }
+ return numSamples - ilength;
+}
+
+
+// Set filter coeffiecients and length.
+//
+// Throws an exception if filter length isn't divisible by 8
+void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint uResultDivFactor)
+{
+ assert(newLength > 0);
+ if (newLength % 8) ST_THROW_RT_ERROR("FIR filter length not divisible by 8");
+
+ #ifdef SOUNDTOUCH_FLOAT_SAMPLES
+ // scale coefficients already here if using floating samples
+ double scale = 1.0 / resultDivider;
+ #else
+ short scale = 1;
+ #endif
+
+ lengthDiv8 = newLength / 8;
+ length = lengthDiv8 * 8;
+ assert(length == newLength);
+
+ resultDivFactor = uResultDivFactor;
+ resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor);
+
+ delete[] filterCoeffs;
+ filterCoeffs = new SAMPLETYPE[length];
+ delete[] filterCoeffsStereo;
+ filterCoeffsStereo = new SAMPLETYPE[length*2];
+ for (uint i = 0; i < length; i ++)
+ {
+ filterCoeffs[i] = (SAMPLETYPE)(coeffs[i] * scale);
+ // create also stereo set of filter coefficients: this allows compiler
+ // to autovectorize filter evaluation much more efficiently
+ filterCoeffsStereo[2 * i] = (SAMPLETYPE)(coeffs[i] * scale);
+ filterCoeffsStereo[2 * i + 1] = (SAMPLETYPE)(coeffs[i] * scale);
+ }
+}
+
+
+uint FIRFilter::getLength() const
+{
+ return length;
+}
+
+
+// Applies the filter to the given sequence of samples.
+//
+// Note : The amount of outputted samples is by value of 'filter_length'
+// smaller than the amount of input samples.
+uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
+{
+ assert(length > 0);
+ assert(lengthDiv8 * 8 == length);
+
+ if (numSamples < length) return 0;
+
+#ifndef USE_MULTICH_ALWAYS
+ if (numChannels == 1)
+ {
+ return evaluateFilterMono(dest, src, numSamples);
+ }
+ else if (numChannels == 2)
+ {
+ return evaluateFilterStereo(dest, src, numSamples);
+ }
+ else
+#endif // USE_MULTICH_ALWAYS
+ {
+ assert(numChannels > 0);
+ return evaluateFilterMulti(dest, src, numSamples, numChannels);
+ }
+}
+
+
+// Operator 'new' is overloaded so that it automatically creates a suitable instance
+// depending on if we've a MMX-capable CPU available or not.
+void * FIRFilter::operator new(size_t s)
+{
+ // Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!
+ ST_THROW_RT_ERROR("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");
+ return newInstance();
+}
+
+
+FIRFilter * FIRFilter::newInstance()
+{
+#if defined(SOUNDTOUCH_ALLOW_MMX) || defined(SOUNDTOUCH_ALLOW_SSE)
+ uint uExtensions;
+
+ uExtensions = detectCPUextensions();
+#endif
+
+ // Check if MMX/SSE instruction set extensions supported by CPU
+
+#ifdef SOUNDTOUCH_ALLOW_MMX
+ // MMX routines available only with integer sample types
+ if (uExtensions & SUPPORT_MMX)
+ {
+ return ::new FIRFilterMMX;
+ }
+ else
+#endif // SOUNDTOUCH_ALLOW_MMX
+
+#ifdef SOUNDTOUCH_ALLOW_SSE
+ if (uExtensions & SUPPORT_SSE)
+ {
+ // SSE support
+ return ::new FIRFilterSSE;
+ }
+ else
+#endif // SOUNDTOUCH_ALLOW_SSE
+
+ {
+ // ISA optimizations not supported, use plain C version
+ return ::new FIRFilter;
+ }
+}
diff --git a/media/libsoundtouch/src/FIRFilter.h b/media/libsoundtouch/src/FIRFilter.h
new file mode 100644
index 0000000000..39c2cc7542
--- /dev/null
+++ b/media/libsoundtouch/src/FIRFilter.h
@@ -0,0 +1,140 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// General FIR digital filter routines with MMX optimization.
+///
+/// Note : MMX optimized functions reside in a separate, platform-specific file,
+/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef FIRFilter_H
+#define FIRFilter_H
+
+#include <stddef.h>
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class FIRFilter
+{
+protected:
+ // Number of FIR filter taps
+ uint length;
+ // Number of FIR filter taps divided by 8
+ uint lengthDiv8;
+
+ // Result divider factor in 2^k format
+ uint resultDivFactor;
+
+ // Result divider value.
+ SAMPLETYPE resultDivider;
+
+ // Memory for filter coefficients
+ SAMPLETYPE *filterCoeffs;
+ SAMPLETYPE *filterCoeffsStereo;
+
+ virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples) const;
+ virtual uint evaluateFilterMono(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples) const;
+ virtual uint evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels);
+
+public:
+ FIRFilter();
+ virtual ~FIRFilter();
+
+ /// Operator 'new' is overloaded so that it automatically creates a suitable instance
+ /// depending on if we've a MMX-capable CPU available or not.
+ static void * operator new(size_t s);
+
+ static FIRFilter *newInstance();
+
+ /// Applies the filter to the given sequence of samples.
+ /// Note : The amount of outputted samples is by value of 'filter_length'
+ /// smaller than the amount of input samples.
+ ///
+ /// \return Number of samples copied to 'dest'.
+ uint evaluate(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ uint numSamples,
+ uint numChannels);
+
+ uint getLength() const;
+
+ virtual void setCoefficients(const SAMPLETYPE *coeffs,
+ uint newLength,
+ uint uResultDivFactor);
+};
+
+
+// Optional subclasses that implement CPU-specific optimizations:
+
+#ifdef SOUNDTOUCH_ALLOW_MMX
+
+/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
+ class FIRFilterMMX : public FIRFilter
+ {
+ protected:
+ short *filterCoeffsUnalign;
+ short *filterCoeffsAlign;
+
+ virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const;
+ public:
+ FIRFilterMMX();
+ ~FIRFilterMMX();
+
+ virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);
+ };
+
+#endif // SOUNDTOUCH_ALLOW_MMX
+
+
+#ifdef SOUNDTOUCH_ALLOW_SSE
+ /// Class that implements SSE optimized functions exclusive for floating point samples type.
+ class FIRFilterSSE : public FIRFilter
+ {
+ protected:
+ float *filterCoeffsUnalign;
+ float *filterCoeffsAlign;
+
+ virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
+ public:
+ FIRFilterSSE();
+ ~FIRFilterSSE();
+
+ virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
+ };
+
+#endif // SOUNDTOUCH_ALLOW_SSE
+
+}
+
+#endif // FIRFilter_H
diff --git a/media/libsoundtouch/src/InterpolateCubic.cpp b/media/libsoundtouch/src/InterpolateCubic.cpp
new file mode 100644
index 0000000000..b37b0fa801
--- /dev/null
+++ b/media/libsoundtouch/src/InterpolateCubic.cpp
@@ -0,0 +1,196 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Cubic interpolation routine.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <stddef.h>
+#include <math.h>
+#include "InterpolateCubic.h"
+#include "STTypes.h"
+
+using namespace soundtouch;
+
+// cubic interpolation coefficients
+static const float _coeffs[]=
+{ -0.5f, 1.0f, -0.5f, 0.0f,
+ 1.5f, -2.5f, 0.0f, 1.0f,
+ -1.5f, 2.0f, 0.5f, 0.0f,
+ 0.5f, -0.5f, 0.0f, 0.0f};
+
+
+InterpolateCubic::InterpolateCubic()
+{
+ fract = 0;
+}
+
+
+void InterpolateCubic::resetRegisters()
+{
+ fract = 0;
+}
+
+
+/// Transpose mono audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateCubic::transposeMono(SAMPLETYPE *pdest,
+ const SAMPLETYPE *psrc,
+ int &srcSamples)
+{
+ int i;
+ int srcSampleEnd = srcSamples - 4;
+ int srcCount = 0;
+
+ i = 0;
+ while (srcCount < srcSampleEnd)
+ {
+ float out;
+ const float x3 = 1.0f;
+ const float x2 = (float)fract; // x
+ const float x1 = x2*x2; // x^2
+ const float x0 = x1*x2; // x^3
+ float y0, y1, y2, y3;
+
+ assert(fract < 1.0);
+
+ y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
+ y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
+ y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
+ y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
+
+ out = y0 * psrc[0] + y1 * psrc[1] + y2 * psrc[2] + y3 * psrc[3];
+
+ pdest[i] = (SAMPLETYPE)out;
+ i ++;
+
+ // update position fraction
+ fract += rate;
+ // update whole positions
+ int whole = (int)fract;
+ fract -= whole;
+ psrc += whole;
+ srcCount += whole;
+ }
+ srcSamples = srcCount;
+ return i;
+}
+
+
+/// Transpose stereo audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateCubic::transposeStereo(SAMPLETYPE *pdest,
+ const SAMPLETYPE *psrc,
+ int &srcSamples)
+{
+ int i;
+ int srcSampleEnd = srcSamples - 4;
+ int srcCount = 0;
+
+ i = 0;
+ while (srcCount < srcSampleEnd)
+ {
+ const float x3 = 1.0f;
+ const float x2 = (float)fract; // x
+ const float x1 = x2*x2; // x^2
+ const float x0 = x1*x2; // x^3
+ float y0, y1, y2, y3;
+ float out0, out1;
+
+ assert(fract < 1.0);
+
+ y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
+ y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
+ y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
+ y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
+
+ out0 = y0 * psrc[0] + y1 * psrc[2] + y2 * psrc[4] + y3 * psrc[6];
+ out1 = y0 * psrc[1] + y1 * psrc[3] + y2 * psrc[5] + y3 * psrc[7];
+
+ pdest[2*i] = (SAMPLETYPE)out0;
+ pdest[2*i+1] = (SAMPLETYPE)out1;
+ i ++;
+
+ // update position fraction
+ fract += rate;
+ // update whole positions
+ int whole = (int)fract;
+ fract -= whole;
+ psrc += 2*whole;
+ srcCount += whole;
+ }
+ srcSamples = srcCount;
+ return i;
+}
+
+
+/// Transpose multi-channel audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateCubic::transposeMulti(SAMPLETYPE *pdest,
+ const SAMPLETYPE *psrc,
+ int &srcSamples)
+{
+ int i;
+ int srcSampleEnd = srcSamples - 4;
+ int srcCount = 0;
+
+ i = 0;
+ while (srcCount < srcSampleEnd)
+ {
+ const float x3 = 1.0f;
+ const float x2 = (float)fract; // x
+ const float x1 = x2*x2; // x^2
+ const float x0 = x1*x2; // x^3
+ float y0, y1, y2, y3;
+
+ assert(fract < 1.0);
+
+ y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
+ y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
+ y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
+ y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
+
+ for (int c = 0; c < numChannels; c ++)
+ {
+ float out;
+ out = y0 * psrc[c] + y1 * psrc[c + numChannels] + y2 * psrc[c + 2 * numChannels] + y3 * psrc[c + 3 * numChannels];
+ pdest[0] = (SAMPLETYPE)out;
+ pdest ++;
+ }
+ i ++;
+
+ // update position fraction
+ fract += rate;
+ // update whole positions
+ int whole = (int)fract;
+ fract -= whole;
+ psrc += numChannels*whole;
+ srcCount += whole;
+ }
+ srcSamples = srcCount;
+ return i;
+}
diff --git a/media/libsoundtouch/src/InterpolateCubic.h b/media/libsoundtouch/src/InterpolateCubic.h
new file mode 100644
index 0000000000..481abd64bc
--- /dev/null
+++ b/media/libsoundtouch/src/InterpolateCubic.h
@@ -0,0 +1,69 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Cubic interpolation routine.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _InterpolateCubic_H_
+#define _InterpolateCubic_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class InterpolateCubic : public TransposerBase
+{
+protected:
+ virtual int transposeMono(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples);
+ virtual int transposeStereo(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples);
+ virtual int transposeMulti(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples);
+
+ double fract;
+
+public:
+ InterpolateCubic();
+
+ virtual void resetRegisters();
+
+ int getLatency() const
+ {
+ return 1;
+ }
+};
+
+}
+
+#endif
diff --git a/media/libsoundtouch/src/InterpolateLinear.cpp b/media/libsoundtouch/src/InterpolateLinear.cpp
new file mode 100644
index 0000000000..9533e79b79
--- /dev/null
+++ b/media/libsoundtouch/src/InterpolateLinear.cpp
@@ -0,0 +1,296 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Linear interpolation algorithm.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <assert.h>
+#include <stdlib.h>
+#include "InterpolateLinear.h"
+
+using namespace soundtouch;
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// InterpolateLinearInteger - integer arithmetic implementation
+//
+
+/// fixed-point interpolation routine precision
+#define SCALE 65536
+
+
+// Constructor
+InterpolateLinearInteger::InterpolateLinearInteger() : TransposerBase()
+{
+ // Notice: use local function calling syntax for sake of clarity,
+ // to indicate the fact that C++ constructor can't call virtual functions.
+ resetRegisters();
+ setRate(1.0f);
+}
+
+
+void InterpolateLinearInteger::resetRegisters()
+{
+ iFract = 0;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation.
+// 'Mono' version of the routine. Returns the number of samples returned in
+// the "dest" buffer
+int InterpolateLinearInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+ int i;
+ int srcSampleEnd = srcSamples - 1;
+ int srcCount = 0;
+
+ i = 0;
+ while (srcCount < srcSampleEnd)
+ {
+ LONG_SAMPLETYPE temp;
+
+ assert(iFract < SCALE);
+
+ temp = (SCALE - iFract) * src[0] + iFract * src[1];
+ dest[i] = (SAMPLETYPE)(temp / SCALE);
+ i++;
+
+ iFract += iRate;
+
+ int iWhole = iFract / SCALE;
+ iFract -= iWhole * SCALE;
+ srcCount += iWhole;
+ src += iWhole;
+ }
+ srcSamples = srcCount;
+
+ return i;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation.
+// 'Stereo' version of the routine. Returns the number of samples returned in
+// the "dest" buffer
+int InterpolateLinearInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+ int i;
+ int srcSampleEnd = srcSamples - 1;
+ int srcCount = 0;
+
+ i = 0;
+ while (srcCount < srcSampleEnd)
+ {
+ LONG_SAMPLETYPE temp0;
+ LONG_SAMPLETYPE temp1;
+
+ assert(iFract < SCALE);
+
+ temp0 = (SCALE - iFract) * src[0] + iFract * src[2];
+ temp1 = (SCALE - iFract) * src[1] + iFract * src[3];
+ dest[0] = (SAMPLETYPE)(temp0 / SCALE);
+ dest[1] = (SAMPLETYPE)(temp1 / SCALE);
+ dest += 2;
+ i++;
+
+ iFract += iRate;
+
+ int iWhole = iFract / SCALE;
+ iFract -= iWhole * SCALE;
+ srcCount += iWhole;
+ src += 2*iWhole;
+ }
+ srcSamples = srcCount;
+
+ return i;
+}
+
+
+int InterpolateLinearInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+ int i;
+ int srcSampleEnd = srcSamples - 1;
+ int srcCount = 0;
+
+ i = 0;
+ while (srcCount < srcSampleEnd)
+ {
+ LONG_SAMPLETYPE temp, vol1;
+
+ assert(iFract < SCALE);
+ vol1 = (LONG_SAMPLETYPE)(SCALE - iFract);
+ for (int c = 0; c < numChannels; c ++)
+ {
+ temp = vol1 * src[c] + iFract * src[c + numChannels];
+ dest[0] = (SAMPLETYPE)(temp / SCALE);
+ dest ++;
+ }
+ i++;
+
+ iFract += iRate;
+
+ int iWhole = iFract / SCALE;
+ iFract -= iWhole * SCALE;
+ srcCount += iWhole;
+ src += iWhole * numChannels;
+ }
+ srcSamples = srcCount;
+
+ return i;
+}
+
+
+// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
+// iRate, larger faster iRates.
+void InterpolateLinearInteger::setRate(double newRate)
+{
+ iRate = (int)(newRate * SCALE + 0.5);
+ TransposerBase::setRate(newRate);
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// InterpolateLinearFloat - floating point arithmetic implementation
+//
+//////////////////////////////////////////////////////////////////////////////
+
+
+// Constructor
+InterpolateLinearFloat::InterpolateLinearFloat() : TransposerBase()
+{
+ // Notice: use local function calling syntax for sake of clarity,
+ // to indicate the fact that C++ constructor can't call virtual functions.
+ resetRegisters();
+ setRate(1.0);
+}
+
+
+void InterpolateLinearFloat::resetRegisters()
+{
+ fract = 0;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation.
+// 'Mono' version of the routine. Returns the number of samples returned in
+// the "dest" buffer
+int InterpolateLinearFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+ int i;
+ int srcSampleEnd = srcSamples - 1;
+ int srcCount = 0;
+
+ i = 0;
+ while (srcCount < srcSampleEnd)
+ {
+ double out;
+ assert(fract < 1.0);
+
+ out = (1.0 - fract) * src[0] + fract * src[1];
+ dest[i] = (SAMPLETYPE)out;
+ i ++;
+
+ // update position fraction
+ fract += rate;
+ // update whole positions
+ int whole = (int)fract;
+ fract -= whole;
+ src += whole;
+ srcCount += whole;
+ }
+ srcSamples = srcCount;
+ return i;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation.
+// 'Mono' version of the routine. Returns the number of samples returned in
+// the "dest" buffer
+int InterpolateLinearFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+ int i;
+ int srcSampleEnd = srcSamples - 1;
+ int srcCount = 0;
+
+ i = 0;
+ while (srcCount < srcSampleEnd)
+ {
+ double out0, out1;
+ assert(fract < 1.0);
+
+ out0 = (1.0 - fract) * src[0] + fract * src[2];
+ out1 = (1.0 - fract) * src[1] + fract * src[3];
+ dest[2*i] = (SAMPLETYPE)out0;
+ dest[2*i+1] = (SAMPLETYPE)out1;
+ i ++;
+
+ // update position fraction
+ fract += rate;
+ // update whole positions
+ int whole = (int)fract;
+ fract -= whole;
+ src += 2*whole;
+ srcCount += whole;
+ }
+ srcSamples = srcCount;
+ return i;
+}
+
+
+int InterpolateLinearFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
+{
+ int i;
+ int srcSampleEnd = srcSamples - 1;
+ int srcCount = 0;
+
+ i = 0;
+ while (srcCount < srcSampleEnd)
+ {
+ float temp, vol1, fract_float;
+
+ vol1 = (float)(1.0 - fract);
+ fract_float = (float)fract;
+ for (int c = 0; c < numChannels; c ++)
+ {
+ temp = vol1 * src[c] + fract_float * src[c + numChannels];
+ *dest = (SAMPLETYPE)temp;
+ dest ++;
+ }
+ i++;
+
+ fract += rate;
+
+ int iWhole = (int)fract;
+ fract -= iWhole;
+ srcCount += iWhole;
+ src += iWhole * numChannels;
+ }
+ srcSamples = srcCount;
+
+ return i;
+}
diff --git a/media/libsoundtouch/src/InterpolateLinear.h b/media/libsoundtouch/src/InterpolateLinear.h
new file mode 100644
index 0000000000..ff362e84b1
--- /dev/null
+++ b/media/libsoundtouch/src/InterpolateLinear.h
@@ -0,0 +1,98 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Linear interpolation routine.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _InterpolateLinear_H_
+#define _InterpolateLinear_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// Linear transposer class that uses integer arithmetic
+class InterpolateLinearInteger : public TransposerBase
+{
+protected:
+ int iFract;
+ int iRate;
+
+ virtual int transposeMono(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples);
+ virtual int transposeStereo(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples);
+ virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
+public:
+ InterpolateLinearInteger();
+
+ /// Sets new target rate. Normal rate = 1.0, smaller values represent slower
+ /// rate, larger faster rates.
+ virtual void setRate(double newRate);
+
+ virtual void resetRegisters();
+
+ int getLatency() const
+ {
+ return 0;
+ }
+};
+
+
+/// Linear transposer class that uses floating point arithmetic
+class InterpolateLinearFloat : public TransposerBase
+{
+protected:
+ double fract;
+
+ virtual int transposeMono(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples);
+ virtual int transposeStereo(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples);
+ virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
+
+public:
+ InterpolateLinearFloat();
+
+ virtual void resetRegisters();
+
+ int getLatency() const
+ {
+ return 0;
+ }
+};
+
+}
+
+#endif
diff --git a/media/libsoundtouch/src/InterpolateShannon.cpp b/media/libsoundtouch/src/InterpolateShannon.cpp
new file mode 100644
index 0000000000..975d872ad6
--- /dev/null
+++ b/media/libsoundtouch/src/InterpolateShannon.cpp
@@ -0,0 +1,181 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Sample interpolation routine using 8-tap band-limited Shannon interpolation
+/// with kaiser window.
+///
+/// Notice. This algorithm is remarkably much heavier than linear or cubic
+/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
+/// for experimental purposes
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <math.h>
+#include "InterpolateShannon.h"
+#include "STTypes.h"
+
+using namespace soundtouch;
+
+
+/// Kaiser window with beta = 2.0
+/// Values scaled down by 5% to avoid overflows
+static const double _kaiser8[8] =
+{
+ 0.41778693317814,
+ 0.64888025049173,
+ 0.83508562409944,
+ 0.93887857733412,
+ 0.93887857733412,
+ 0.83508562409944,
+ 0.64888025049173,
+ 0.41778693317814
+};
+
+
+InterpolateShannon::InterpolateShannon()
+{
+ fract = 0;
+}
+
+
+void InterpolateShannon::resetRegisters()
+{
+ fract = 0;
+}
+
+
+#define PI 3.1415926536
+#define sinc(x) (sin(PI * (x)) / (PI * (x)))
+
+/// Transpose mono audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateShannon::transposeMono(SAMPLETYPE *pdest,
+ const SAMPLETYPE *psrc,
+ int &srcSamples)
+{
+ int i;
+ int srcSampleEnd = srcSamples - 8;
+ int srcCount = 0;
+
+ i = 0;
+ while (srcCount < srcSampleEnd)
+ {
+ double out;
+ assert(fract < 1.0);
+
+ out = psrc[0] * sinc(-3.0 - fract) * _kaiser8[0];
+ out += psrc[1] * sinc(-2.0 - fract) * _kaiser8[1];
+ out += psrc[2] * sinc(-1.0 - fract) * _kaiser8[2];
+ if (fract < 1e-6)
+ {
+ out += psrc[3] * _kaiser8[3]; // sinc(0) = 1
+ }
+ else
+ {
+ out += psrc[3] * sinc(- fract) * _kaiser8[3];
+ }
+ out += psrc[4] * sinc( 1.0 - fract) * _kaiser8[4];
+ out += psrc[5] * sinc( 2.0 - fract) * _kaiser8[5];
+ out += psrc[6] * sinc( 3.0 - fract) * _kaiser8[6];
+ out += psrc[7] * sinc( 4.0 - fract) * _kaiser8[7];
+
+ pdest[i] = (SAMPLETYPE)out;
+ i ++;
+
+ // update position fraction
+ fract += rate;
+ // update whole positions
+ int whole = (int)fract;
+ fract -= whole;
+ psrc += whole;
+ srcCount += whole;
+ }
+ srcSamples = srcCount;
+ return i;
+}
+
+
+/// Transpose stereo audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateShannon::transposeStereo(SAMPLETYPE *pdest,
+ const SAMPLETYPE *psrc,
+ int &srcSamples)
+{
+ int i;
+ int srcSampleEnd = srcSamples - 8;
+ int srcCount = 0;
+
+ i = 0;
+ while (srcCount < srcSampleEnd)
+ {
+ double out0, out1, w;
+ assert(fract < 1.0);
+
+ w = sinc(-3.0 - fract) * _kaiser8[0];
+ out0 = psrc[0] * w; out1 = psrc[1] * w;
+ w = sinc(-2.0 - fract) * _kaiser8[1];
+ out0 += psrc[2] * w; out1 += psrc[3] * w;
+ w = sinc(-1.0 - fract) * _kaiser8[2];
+ out0 += psrc[4] * w; out1 += psrc[5] * w;
+ w = _kaiser8[3] * ((fract < 1e-5) ? 1.0 : sinc(- fract)); // sinc(0) = 1
+ out0 += psrc[6] * w; out1 += psrc[7] * w;
+ w = sinc( 1.0 - fract) * _kaiser8[4];
+ out0 += psrc[8] * w; out1 += psrc[9] * w;
+ w = sinc( 2.0 - fract) * _kaiser8[5];
+ out0 += psrc[10] * w; out1 += psrc[11] * w;
+ w = sinc( 3.0 - fract) * _kaiser8[6];
+ out0 += psrc[12] * w; out1 += psrc[13] * w;
+ w = sinc( 4.0 - fract) * _kaiser8[7];
+ out0 += psrc[14] * w; out1 += psrc[15] * w;
+
+ pdest[2*i] = (SAMPLETYPE)out0;
+ pdest[2*i+1] = (SAMPLETYPE)out1;
+ i ++;
+
+ // update position fraction
+ fract += rate;
+ // update whole positions
+ int whole = (int)fract;
+ fract -= whole;
+ psrc += 2*whole;
+ srcCount += whole;
+ }
+ srcSamples = srcCount;
+ return i;
+}
+
+
+/// Transpose stereo audio. Returns number of produced output samples, and
+/// updates "srcSamples" to amount of consumed source samples
+int InterpolateShannon::transposeMulti(SAMPLETYPE *pdest,
+ const SAMPLETYPE *psrc,
+ int &srcSamples)
+{
+ // not implemented
+ assert(false);
+ return 0;
+}
diff --git a/media/libsoundtouch/src/InterpolateShannon.h b/media/libsoundtouch/src/InterpolateShannon.h
new file mode 100644
index 0000000000..72ab0b526d
--- /dev/null
+++ b/media/libsoundtouch/src/InterpolateShannon.h
@@ -0,0 +1,74 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Sample interpolation routine using 8-tap band-limited Shannon interpolation
+/// with kaiser window.
+///
+/// Notice. This algorithm is remarkably much heavier than linear or cubic
+/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
+/// for experimental purposes
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _InterpolateShannon_H_
+#define _InterpolateShannon_H_
+
+#include "RateTransposer.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+class InterpolateShannon : public TransposerBase
+{
+protected:
+ int transposeMono(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples);
+ int transposeStereo(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples);
+ int transposeMulti(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples);
+
+ double fract;
+
+public:
+ InterpolateShannon();
+
+ void resetRegisters();
+
+ int getLatency() const
+ {
+ return 3;
+ }
+};
+
+}
+
+#endif
diff --git a/media/libsoundtouch/src/RateTransposer.cpp b/media/libsoundtouch/src/RateTransposer.cpp
new file mode 100644
index 0000000000..4c202391e0
--- /dev/null
+++ b/media/libsoundtouch/src/RateTransposer.cpp
@@ -0,0 +1,315 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Sample rate transposer. Changes sample rate by using linear interpolation
+/// together with anti-alias filtering (first order interpolation with anti-
+/// alias filtering should be quite adequate for this application)
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <memory.h>
+#include <assert.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include "RateTransposer.h"
+#include "InterpolateLinear.h"
+#include "InterpolateCubic.h"
+#include "InterpolateShannon.h"
+#include "AAFilter.h"
+
+using namespace soundtouch;
+
+// Define default interpolation algorithm here
+TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
+
+
+// Constructor
+RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
+{
+ bUseAAFilter =
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+ true;
+#else
+ // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
+ false;
+#endif
+
+ // Instantiates the anti-alias filter
+ pAAFilter = new AAFilter(64);
+ pTransposer = TransposerBase::newInstance();
+ clear();
+}
+
+
+RateTransposer::~RateTransposer()
+{
+ delete pAAFilter;
+ delete pTransposer;
+}
+
+
+/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
+void RateTransposer::enableAAFilter(bool newMode)
+{
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+ // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
+ bUseAAFilter = newMode;
+ clear();
+#endif
+}
+
+
+/// Returns nonzero if anti-alias filter is enabled.
+bool RateTransposer::isAAFilterEnabled() const
+{
+ return bUseAAFilter;
+}
+
+
+AAFilter *RateTransposer::getAAFilter()
+{
+ return pAAFilter;
+}
+
+
+// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
+// iRate, larger faster iRates.
+void RateTransposer::setRate(double newRate)
+{
+ double fCutoff;
+
+ pTransposer->setRate(newRate);
+
+ // design a new anti-alias filter
+ if (newRate > 1.0)
+ {
+ fCutoff = 0.5 / newRate;
+ }
+ else
+ {
+ fCutoff = 0.5 * newRate;
+ }
+ pAAFilter->setCutoffFreq(fCutoff);
+}
+
+
+// Adds 'nSamples' pcs of samples from the 'samples' memory position into
+// the input of the object.
+void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
+{
+ processSamples(samples, nSamples);
+}
+
+
+// Transposes sample rate by applying anti-alias filter to prevent folding.
+// Returns amount of samples returned in the "dest" buffer.
+// The maximum amount of samples that can be returned at a time is set by
+// the 'set_returnBuffer_size' function.
+void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
+{
+ uint count;
+
+ if (nSamples == 0) return;
+
+ // Store samples to input buffer
+ inputBuffer.putSamples(src, nSamples);
+
+ // If anti-alias filter is turned off, simply transpose without applying
+ // the filter
+ if (bUseAAFilter == false)
+ {
+ count = pTransposer->transpose(outputBuffer, inputBuffer);
+ return;
+ }
+
+ assert(pAAFilter);
+
+ // Transpose with anti-alias filter
+ if (pTransposer->rate < 1.0f)
+ {
+ // If the parameter 'Rate' value is smaller than 1, first transpose
+ // the samples and then apply the anti-alias filter to remove aliasing.
+
+ // Transpose the samples, store the result to end of "midBuffer"
+ pTransposer->transpose(midBuffer, inputBuffer);
+
+ // Apply the anti-alias filter for transposed samples in midBuffer
+ pAAFilter->evaluate(outputBuffer, midBuffer);
+ }
+ else
+ {
+ // If the parameter 'Rate' value is larger than 1, first apply the
+ // anti-alias filter to remove high frequencies (prevent them from folding
+ // over the lover frequencies), then transpose.
+
+ // Apply the anti-alias filter for samples in inputBuffer
+ pAAFilter->evaluate(midBuffer, inputBuffer);
+
+ // Transpose the AA-filtered samples in "midBuffer"
+ pTransposer->transpose(outputBuffer, midBuffer);
+ }
+}
+
+
+// Sets the number of channels, 1 = mono, 2 = stereo
+void RateTransposer::setChannels(int nChannels)
+{
+ if (!verifyNumberOfChannels(nChannels) ||
+ (pTransposer->numChannels == nChannels)) return;
+
+ pTransposer->setChannels(nChannels);
+ inputBuffer.setChannels(nChannels);
+ midBuffer.setChannels(nChannels);
+ outputBuffer.setChannels(nChannels);
+}
+
+
+// Clears all the samples in the object
+void RateTransposer::clear()
+{
+ outputBuffer.clear();
+ midBuffer.clear();
+ inputBuffer.clear();
+ pTransposer->resetRegisters();
+
+ // prefill buffer to avoid losing first samples at beginning of stream
+ int prefill = getLatency();
+ inputBuffer.addSilent(prefill);
+}
+
+
+// Returns nonzero if there aren't any samples available for outputting.
+int RateTransposer::isEmpty() const
+{
+ int res;
+
+ res = FIFOProcessor::isEmpty();
+ if (res == 0) return 0;
+ return inputBuffer.isEmpty();
+}
+
+
+/// Return approximate initial input-output latency
+int RateTransposer::getLatency() const
+{
+ return pTransposer->getLatency() +
+ ((bUseAAFilter) ? (pAAFilter->getLength() / 2) : 0);
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// TransposerBase - Base class for interpolation
+//
+
+// static function to set interpolation algorithm
+void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
+{
+ TransposerBase::algorithm = a;
+}
+
+
+// Transposes the sample rate of the given samples using linear interpolation.
+// Returns the number of samples returned in the "dest" buffer
+int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
+{
+ int numSrcSamples = src.numSamples();
+ int sizeDemand = (int)((double)numSrcSamples / rate) + 8;
+ int numOutput;
+ SAMPLETYPE *psrc = src.ptrBegin();
+ SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
+
+#ifndef USE_MULTICH_ALWAYS
+ if (numChannels == 1)
+ {
+ numOutput = transposeMono(pdest, psrc, numSrcSamples);
+ }
+ else if (numChannels == 2)
+ {
+ numOutput = transposeStereo(pdest, psrc, numSrcSamples);
+ }
+ else
+#endif // USE_MULTICH_ALWAYS
+ {
+ assert(numChannels > 0);
+ numOutput = transposeMulti(pdest, psrc, numSrcSamples);
+ }
+ dest.putSamples(numOutput);
+ src.receiveSamples(numSrcSamples);
+ return numOutput;
+}
+
+
+TransposerBase::TransposerBase()
+{
+ numChannels = 0;
+ rate = 1.0f;
+}
+
+
+TransposerBase::~TransposerBase()
+{
+}
+
+
+void TransposerBase::setChannels(int channels)
+{
+ numChannels = channels;
+ resetRegisters();
+}
+
+
+void TransposerBase::setRate(double newRate)
+{
+ rate = newRate;
+}
+
+
+// static factory function
+TransposerBase *TransposerBase::newInstance()
+{
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+ // Notice: For integer arithmetic support only linear algorithm (due to simplest calculus)
+ return ::new InterpolateLinearInteger;
+#else
+ switch (algorithm)
+ {
+ case LINEAR:
+ return new InterpolateLinearFloat;
+
+ case CUBIC:
+ return new InterpolateCubic;
+
+ case SHANNON:
+ return new InterpolateShannon;
+
+ default:
+ assert(false);
+ return NULL;
+ }
+#endif
+}
diff --git a/media/libsoundtouch/src/RateTransposer.h b/media/libsoundtouch/src/RateTransposer.h
new file mode 100644
index 0000000000..59381fab5f
--- /dev/null
+++ b/media/libsoundtouch/src/RateTransposer.h
@@ -0,0 +1,164 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Sample rate transposer. Changes sample rate by using linear interpolation
+/// together with anti-alias filtering (first order interpolation with anti-
+/// alias filtering should be quite adequate for this application).
+///
+/// Use either of the derived classes of 'RateTransposerInteger' or
+/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
+/// algorithm implementation.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef RateTransposer_H
+#define RateTransposer_H
+
+#include <stddef.h>
+#include "AAFilter.h"
+#include "FIFOSamplePipe.h"
+#include "FIFOSampleBuffer.h"
+
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// Abstract base class for transposer implementations (linear, advanced vs integer, float etc)
+class TransposerBase
+{
+public:
+ enum ALGORITHM {
+ LINEAR = 0,
+ CUBIC,
+ SHANNON
+ };
+
+protected:
+ virtual int transposeMono(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples) = 0;
+ virtual int transposeStereo(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples) = 0;
+ virtual int transposeMulti(SAMPLETYPE *dest,
+ const SAMPLETYPE *src,
+ int &srcSamples) = 0;
+
+ static ALGORITHM algorithm;
+
+public:
+ double rate;
+ int numChannels;
+
+ TransposerBase();
+ virtual ~TransposerBase();
+
+ virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
+ virtual void setRate(double newRate);
+ virtual void setChannels(int channels);
+ virtual int getLatency() const = 0;
+
+ virtual void resetRegisters() = 0;
+
+ // static factory function
+ static TransposerBase *newInstance();
+
+ // static function to set interpolation algorithm
+ static void setAlgorithm(ALGORITHM a);
+};
+
+
+/// A common linear samplerate transposer class.
+///
+class RateTransposer : public FIFOProcessor
+{
+protected:
+ /// Anti-alias filter object
+ AAFilter *pAAFilter;
+ TransposerBase *pTransposer;
+
+ /// Buffer for collecting samples to feed the anti-alias filter between
+ /// two batches
+ FIFOSampleBuffer inputBuffer;
+
+ /// Buffer for keeping samples between transposing & anti-alias filter
+ FIFOSampleBuffer midBuffer;
+
+ /// Output sample buffer
+ FIFOSampleBuffer outputBuffer;
+
+ bool bUseAAFilter;
+
+
+ /// Transposes sample rate by applying anti-alias filter to prevent folding.
+ /// Returns amount of samples returned in the "dest" buffer.
+ /// The maximum amount of samples that can be returned at a time is set by
+ /// the 'set_returnBuffer_size' function.
+ void processSamples(const SAMPLETYPE *src,
+ uint numSamples);
+
+public:
+ RateTransposer();
+ virtual ~RateTransposer();
+
+ /// Returns the output buffer object
+ FIFOSamplePipe *getOutput() { return &outputBuffer; };
+
+ /// Return anti-alias filter object
+ AAFilter *getAAFilter();
+
+ /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
+ void enableAAFilter(bool newMode);
+
+ /// Returns nonzero if anti-alias filter is enabled.
+ bool isAAFilterEnabled() const;
+
+ /// Sets new target rate. Normal rate = 1.0, smaller values represent slower
+ /// rate, larger faster rates.
+ virtual void setRate(double newRate);
+
+ /// Sets the number of channels, 1 = mono, 2 = stereo
+ void setChannels(int channels);
+
+ /// Adds 'numSamples' pcs of samples from the 'samples' memory position into
+ /// the input of the object.
+ void putSamples(const SAMPLETYPE *samples, uint numSamples);
+
+ /// Clears all the samples in the object
+ void clear();
+
+ /// Returns nonzero if there aren't any samples available for outputting.
+ int isEmpty() const;
+
+ /// Return approximate initial input-output latency
+ int getLatency() const;
+};
+
+}
+
+#endif
diff --git a/media/libsoundtouch/src/STTypes.h b/media/libsoundtouch/src/STTypes.h
new file mode 100644
index 0000000000..d80bc2899e
--- /dev/null
+++ b/media/libsoundtouch/src/STTypes.h
@@ -0,0 +1,195 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Common type definitions for SoundTouch audio processing library.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef STTypes_H
+#define STTypes_H
+
+typedef unsigned int uint;
+typedef unsigned long ulong;
+
+// Patch for MinGW: on Win64 long is 32-bit
+#ifdef _WIN64
+ typedef unsigned long long ulongptr;
+#else
+ typedef ulong ulongptr;
+#endif
+
+
+// Helper macro for aligning pointer up to next 16-byte boundary
+#define SOUNDTOUCH_ALIGN_POINTER_16(x) ( ( (ulongptr)(x) + 15 ) & ~(ulongptr)15 )
+
+
+#include "soundtouch_config.h"
+
+#if defined(WIN32)
+#if defined(BUILDING_SOUNDTOUCH)
+#define SOUNDTOUCH_API __declspec(dllexport)
+#else
+#define SOUNDTOUCH_API __declspec(dllimport)
+#endif
+#else
+#define SOUNDTOUCH_API
+#endif
+
+namespace soundtouch
+{
+ /// Max allowed number of channels
+ #define SOUNDTOUCH_MAX_CHANNELS 16
+
+ /// Activate these undef's to overrule the possible sampletype
+ /// setting inherited from some other header file:
+ //#undef SOUNDTOUCH_INTEGER_SAMPLES
+ //#undef SOUNDTOUCH_FLOAT_SAMPLES
+
+ /// If following flag is defined, always uses multichannel processing
+ /// routines also for mono and stero sound. This is for routine testing
+ /// purposes; output should be same with either routines, yet disabling
+ /// the dedicated mono/stereo processing routines will result in slower
+ /// runtime performance so recommendation is to keep this off.
+ // #define USE_MULTICH_ALWAYS
+
+ #if (defined(__SOFTFP__) && defined(ANDROID))
+ // For Android compilation: Force use of Integer samples in case that
+ // compilation uses soft-floating point emulation - soft-fp is way too slow
+ #undef SOUNDTOUCH_FLOAT_SAMPLES
+ #define SOUNDTOUCH_INTEGER_SAMPLES 1
+ #endif
+
+ #if !(SOUNDTOUCH_INTEGER_SAMPLES || SOUNDTOUCH_FLOAT_SAMPLES)
+
+ /// Choose either 32bit floating point or 16bit integer sampletype
+ /// by choosing one of the following defines, unless this selection
+ /// has already been done in some other file.
+ ////
+ /// Notes:
+ /// - In Windows environment, choose the sample format with the
+ /// following defines.
+ /// - In GNU environment, the floating point samples are used by
+ /// default, but integer samples can be chosen by giving the
+ /// following switch to the configure script:
+ /// ./configure --enable-integer-samples
+ /// However, if you still prefer to select the sample format here
+ /// also in GNU environment, then please #undef the INTEGER_SAMPLE
+ /// and FLOAT_SAMPLE defines first as in comments above.
+ //#define SOUNDTOUCH_INTEGER_SAMPLES 1 //< 16bit integer samples
+ #define SOUNDTOUCH_FLOAT_SAMPLES 1 //< 32bit float samples
+
+ #endif
+
+ #if (_M_IX86 || __i386__ || __x86_64__ || _M_X64)
+ /// Define this to allow X86-specific assembler/intrinsic optimizations.
+ /// Notice that library contains also usual C++ versions of each of these
+ /// these routines, so if you're having difficulties getting the optimized
+ /// routines compiled for whatever reason, you may disable these optimizations
+ /// to make the library compile.
+
+ #define SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS 1
+
+ /// In GNU environment, allow the user to override this setting by
+ /// giving the following switch to the configure script:
+ /// ./configure --disable-x86-optimizations
+ /// ./configure --enable-x86-optimizations=no
+ #ifdef SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS
+ #undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
+ #endif
+ #else
+ /// Always disable optimizations when not using a x86 systems.
+ #undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
+
+ #endif
+
+ // If defined, allows the SIMD-optimized routines to skip unevenly aligned
+ // memory offsets that can cause performance penalty in some SIMD implementations.
+ // Causes slight compromise in sound quality.
+ // #define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
+
+
+ #ifdef SOUNDTOUCH_INTEGER_SAMPLES
+ // 16bit integer sample type
+ typedef short SAMPLETYPE;
+ // data type for sample accumulation: Use 32bit integer to prevent overflows
+ typedef long LONG_SAMPLETYPE;
+
+ #ifdef SOUNDTOUCH_FLOAT_SAMPLES
+ // check that only one sample type is defined
+ #error "conflicting sample types defined"
+ #endif // SOUNDTOUCH_FLOAT_SAMPLES
+
+ #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
+ // Allow MMX optimizations (not available in X64 mode)
+ #if (!_M_X64)
+ #define SOUNDTOUCH_ALLOW_MMX 1
+ #endif
+ #endif
+
+ #else
+
+ // floating point samples
+ typedef float SAMPLETYPE;
+ // data type for sample accumulation: Use float also here to enable
+ // efficient autovectorization
+ typedef float LONG_SAMPLETYPE;
+
+ #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
+ // Allow SSE optimizations
+ #define SOUNDTOUCH_ALLOW_SSE 1
+ #endif
+
+ #endif // SOUNDTOUCH_INTEGER_SAMPLES
+
+ #if ((SOUNDTOUCH_ALLOW_SSE) || (__SSE__) || (SOUNDTOUCH_USE_NEON))
+ #if SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
+ #define ST_SIMD_AVOID_UNALIGNED
+ #endif
+ #endif
+
+};
+
+// define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions:
+// #define ST_NO_EXCEPTION_HANDLING 1
+#ifdef ST_NO_EXCEPTION_HANDLING
+ // Exceptions disabled. Throw asserts instead if enabled.
+ #include <assert.h>
+ #define ST_THROW_RT_ERROR(x) {assert((const char *)x);}
+#else
+ // use c++ standard exceptions
+ #include <stdexcept>
+ #include <string>
+ #define ST_THROW_RT_ERROR(x) {throw std::runtime_error(x);}
+#endif
+
+// When this #define is active, eliminates a clicking sound when the "rate" or "pitch"
+// parameter setting crosses from value <1 to >=1 or vice versa during processing.
+// Default is off as such crossover is untypical case and involves a slight sound
+// quality compromise.
+//#define SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER 1
+
+#endif
diff --git a/media/libsoundtouch/src/SoundTouch.cpp b/media/libsoundtouch/src/SoundTouch.cpp
new file mode 100644
index 0000000000..69fba8b9b5
--- /dev/null
+++ b/media/libsoundtouch/src/SoundTouch.cpp
@@ -0,0 +1,538 @@
+//////////////////////////////////////////////////////////////////////////////
+///
+/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
+///
+/// Notes:
+/// - Initialize the SoundTouch object instance by setting up the sound stream
+/// parameters with functions 'setSampleRate' and 'setChannels', then set
+/// desired tempo/pitch/rate settings with the corresponding functions.
+///
+/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
+/// samples that are to be processed are fed into one of the pipe by calling
+/// function 'putSamples', while the ready processed samples can be read
+/// from the other end of the pipeline with function 'receiveSamples'.
+///
+/// - The SoundTouch processing classes require certain sized 'batches' of
+/// samples in order to process the sound. For this reason the classes buffer
+/// incoming samples until there are enough of samples available for
+/// processing, then they carry out the processing step and consequently
+/// make the processed samples available for outputting.
+///
+/// - For the above reason, the processing routines introduce a certain
+/// 'latency' between the input and output, so that the samples input to
+/// SoundTouch may not be immediately available in the output, and neither
+/// the amount of outputtable samples may not immediately be in direct
+/// relationship with the amount of previously input samples.
+///
+/// - The tempo/pitch/rate control parameters can be altered during processing.
+/// Please notice though that they aren't currently protected by semaphores,
+/// so in multi-thread application external semaphore protection may be
+/// required.
+///
+/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
+/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
+/// tempo and pitch in the same ratio) of the sound. The third available control
+/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
+/// combining the two other controls.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <assert.h>
+#include <stdlib.h>
+#include <memory.h>
+#include <math.h>
+#include <stdio.h>
+
+#include "SoundTouch.h"
+#include "TDStretch.h"
+#include "RateTransposer.h"
+#include "cpu_detect.h"
+
+using namespace soundtouch;
+
+/// test if two floating point numbers are equal
+#define TEST_FLOAT_EQUAL(a, b) (fabs(a - b) < 1e-10)
+
+
+/// Print library version string for autoconf
+extern "C" void soundtouch_ac_test()
+{
+ printf("SoundTouch Version: %s\n",SOUNDTOUCH_VERSION);
+}
+
+
+SoundTouch::SoundTouch()
+{
+ // Initialize rate transposer and tempo changer instances
+
+ pRateTransposer = new RateTransposer();
+ pTDStretch = TDStretch::newInstance();
+
+ setOutPipe(pTDStretch);
+
+ rate = tempo = 0;
+
+ virtualPitch =
+ virtualRate =
+ virtualTempo = 1.0;
+
+ calcEffectiveRateAndTempo();
+
+ samplesExpectedOut = 0;
+ samplesOutput = 0;
+
+ channels = 0;
+ bSrateSet = false;
+}
+
+
+SoundTouch::~SoundTouch()
+{
+ delete pRateTransposer;
+ delete pTDStretch;
+}
+
+
+/// Get SoundTouch library version string
+const char *SoundTouch::getVersionString()
+{
+ static const char *_version = SOUNDTOUCH_VERSION;
+
+ return _version;
+}
+
+
+/// Get SoundTouch library version Id
+uint SoundTouch::getVersionId()
+{
+ return SOUNDTOUCH_VERSION_ID;
+}
+
+
+// Sets the number of channels, 1 = mono, 2 = stereo
+void SoundTouch::setChannels(uint numChannels)
+{
+ if (!verifyNumberOfChannels(numChannels)) return;
+
+ channels = numChannels;
+ pRateTransposer->setChannels((int)numChannels);
+ pTDStretch->setChannels((int)numChannels);
+}
+
+
+// Sets new rate control value. Normal rate = 1.0, smaller values
+// represent slower rate, larger faster rates.
+void SoundTouch::setRate(double newRate)
+{
+ virtualRate = newRate;
+ calcEffectiveRateAndTempo();
+}
+
+
+// Sets new rate control value as a difference in percents compared
+// to the original rate (-50 .. +100 %)
+void SoundTouch::setRateChange(double newRate)
+{
+ virtualRate = 1.0 + 0.01 * newRate;
+ calcEffectiveRateAndTempo();
+}
+
+
+// Sets new tempo control value. Normal tempo = 1.0, smaller values
+// represent slower tempo, larger faster tempo.
+void SoundTouch::setTempo(double newTempo)
+{
+ virtualTempo = newTempo;
+ calcEffectiveRateAndTempo();
+}
+
+
+// Sets new tempo control value as a difference in percents compared
+// to the original tempo (-50 .. +100 %)
+void SoundTouch::setTempoChange(double newTempo)
+{
+ virtualTempo = 1.0 + 0.01 * newTempo;
+ calcEffectiveRateAndTempo();
+}
+
+
+// Sets new pitch control value. Original pitch = 1.0, smaller values
+// represent lower pitches, larger values higher pitch.
+void SoundTouch::setPitch(double newPitch)
+{
+ virtualPitch = newPitch;
+ calcEffectiveRateAndTempo();
+}
+
+
+// Sets pitch change in octaves compared to the original pitch
+// (-1.00 .. +1.00)
+void SoundTouch::setPitchOctaves(double newPitch)
+{
+ virtualPitch = exp(0.69314718056 * newPitch);
+ calcEffectiveRateAndTempo();
+}
+
+
+// Sets pitch change in semi-tones compared to the original pitch
+// (-12 .. +12)
+void SoundTouch::setPitchSemiTones(int newPitch)
+{
+ setPitchOctaves((double)newPitch / 12.0);
+}
+
+
+void SoundTouch::setPitchSemiTones(double newPitch)
+{
+ setPitchOctaves(newPitch / 12.0);
+}
+
+
+// Calculates 'effective' rate and tempo values from the
+// nominal control values.
+void SoundTouch::calcEffectiveRateAndTempo()
+{
+ double oldTempo = tempo;
+ double oldRate = rate;
+
+ tempo = virtualTempo / virtualPitch;
+ rate = virtualPitch * virtualRate;
+
+ if (!TEST_FLOAT_EQUAL(rate,oldRate)) pRateTransposer->setRate(rate);
+ if (!TEST_FLOAT_EQUAL(tempo, oldTempo)) pTDStretch->setTempo(tempo);
+
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+ if (rate <= 1.0f)
+ {
+ if (output != pTDStretch)
+ {
+ FIFOSamplePipe *tempoOut;
+
+ assert(output == pRateTransposer);
+ // move samples in the current output buffer to the output of pTDStretch
+ tempoOut = pTDStretch->getOutput();
+ tempoOut->moveSamples(*output);
+ // move samples in pitch transposer's store buffer to tempo changer's input
+ // deprecated : pTDStretch->moveSamples(*pRateTransposer->getStore());
+
+ output = pTDStretch;
+ }
+ }
+ else
+#endif
+ {
+ if (output != pRateTransposer)
+ {
+ FIFOSamplePipe *transOut;
+
+ assert(output == pTDStretch);
+ // move samples in the current output buffer to the output of pRateTransposer
+ transOut = pRateTransposer->getOutput();
+ transOut->moveSamples(*output);
+ // move samples in tempo changer's input to pitch transposer's input
+ pRateTransposer->moveSamples(*pTDStretch->getInput());
+
+ output = pRateTransposer;
+ }
+ }
+}
+
+
+// Sets sample rate.
+void SoundTouch::setSampleRate(uint srate)
+{
+ // set sample rate, leave other tempo changer parameters as they are.
+ pTDStretch->setParameters((int)srate);
+ bSrateSet = true;
+}
+
+
+// Adds 'numSamples' pcs of samples from the 'samples' memory position into
+// the input of the object.
+void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
+{
+ if (bSrateSet == false)
+ {
+ ST_THROW_RT_ERROR("SoundTouch : Sample rate not defined");
+ }
+ else if (channels == 0)
+ {
+ ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined");
+ }
+
+ // accumulate how many samples are expected out from processing, given the current
+ // processing setting
+ samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo);
+
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+ if (rate <= 1.0f)
+ {
+ // transpose the rate down, output the transposed sound to tempo changer buffer
+ assert(output == pTDStretch);
+ pRateTransposer->putSamples(samples, nSamples);
+ pTDStretch->moveSamples(*pRateTransposer);
+ }
+ else
+#endif
+ {
+ // evaluate the tempo changer, then transpose the rate up,
+ assert(output == pRateTransposer);
+ pTDStretch->putSamples(samples, nSamples);
+ pRateTransposer->moveSamples(*pTDStretch);
+ }
+}
+
+
+// Flushes the last samples from the processing pipeline to the output.
+// Clears also the internal processing buffers.
+//
+// Note: This function is meant for extracting the last samples of a sound
+// stream. This function may introduce additional blank samples in the end
+// of the sound stream, and thus it's not recommended to call this function
+// in the middle of a sound stream.
+void SoundTouch::flush()
+{
+ int i;
+ int numStillExpected;
+ SAMPLETYPE *buff = new SAMPLETYPE[128 * channels];
+
+ // how many samples are still expected to output
+ numStillExpected = (int)((long)(samplesExpectedOut + 0.5) - samplesOutput);
+ if (numStillExpected < 0) numStillExpected = 0;
+
+ memset(buff, 0, 128 * channels * sizeof(SAMPLETYPE));
+ // "Push" the last active samples out from the processing pipeline by
+ // feeding blank samples into the processing pipeline until new,
+ // processed samples appear in the output (not however, more than
+ // 24ksamples in any case)
+ for (i = 0; (numStillExpected > (int)numSamples()) && (i < 200); i ++)
+ {
+ putSamples(buff, 128);
+ }
+
+ adjustAmountOfSamples(numStillExpected);
+
+ delete[] buff;
+
+ // Clear input buffers
+ pTDStretch->clearInput();
+ // yet leave the output intouched as that's where the
+ // flushed samples are!
+}
+
+
+// Changes a setting controlling the processing system behaviour. See the
+// 'SETTING_...' defines for available setting ID's.
+bool SoundTouch::setSetting(int settingId, int value)
+{
+ int sampleRate, sequenceMs, seekWindowMs, overlapMs;
+
+ // read current tdstretch routine parameters
+ pTDStretch->getParameters(&sampleRate, &sequenceMs, &seekWindowMs, &overlapMs);
+
+ switch (settingId)
+ {
+ case SETTING_USE_AA_FILTER :
+ // enables / disabless anti-alias filter
+ pRateTransposer->enableAAFilter((value != 0) ? true : false);
+ return true;
+
+ case SETTING_AA_FILTER_LENGTH :
+ // sets anti-alias filter length
+ pRateTransposer->getAAFilter()->setLength(value);
+ return true;
+
+ case SETTING_USE_QUICKSEEK :
+ // enables / disables tempo routine quick seeking algorithm
+ pTDStretch->enableQuickSeek((value != 0) ? true : false);
+ return true;
+
+ case SETTING_SEQUENCE_MS:
+ // change time-stretch sequence duration parameter
+ pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs);
+ return true;
+
+ case SETTING_SEEKWINDOW_MS:
+ // change time-stretch seek window length parameter
+ pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs);
+ return true;
+
+ case SETTING_OVERLAP_MS:
+ // change time-stretch overlap length parameter
+ pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value);
+ return true;
+
+ default :
+ return false;
+ }
+}
+
+
+// Reads a setting controlling the processing system behaviour. See the
+// 'SETTING_...' defines for available setting ID's.
+//
+// Returns the setting value.
+int SoundTouch::getSetting(int settingId) const
+{
+ int temp;
+
+ switch (settingId)
+ {
+ case SETTING_USE_AA_FILTER :
+ return (uint)pRateTransposer->isAAFilterEnabled();
+
+ case SETTING_AA_FILTER_LENGTH :
+ return pRateTransposer->getAAFilter()->getLength();
+
+ case SETTING_USE_QUICKSEEK :
+ return (uint)pTDStretch->isQuickSeekEnabled();
+
+ case SETTING_SEQUENCE_MS:
+ pTDStretch->getParameters(NULL, &temp, NULL, NULL);
+ return temp;
+
+ case SETTING_SEEKWINDOW_MS:
+ pTDStretch->getParameters(NULL, NULL, &temp, NULL);
+ return temp;
+
+ case SETTING_OVERLAP_MS:
+ pTDStretch->getParameters(NULL, NULL, NULL, &temp);
+ return temp;
+
+ case SETTING_NOMINAL_INPUT_SEQUENCE :
+ {
+ int size = pTDStretch->getInputSampleReq();
+
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+ if (rate <= 1.0)
+ {
+ // transposing done before timestretch, which impacts latency
+ return (int)(size * rate + 0.5);
+ }
+#endif
+ return size;
+ }
+
+ case SETTING_NOMINAL_OUTPUT_SEQUENCE :
+ {
+ int size = pTDStretch->getOutputBatchSize();
+
+ if (rate > 1.0)
+ {
+ // transposing done after timestretch, which impacts latency
+ return (int)(size / rate + 0.5);
+ }
+ return size;
+ }
+
+ case SETTING_INITIAL_LATENCY:
+ {
+ double latency = pTDStretch->getLatency();
+ int latency_tr = pRateTransposer->getLatency();
+
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+ if (rate <= 1.0)
+ {
+ // transposing done before timestretch, which impacts latency
+ latency = (latency + latency_tr) * rate;
+ }
+ else
+#endif
+ {
+ latency += (double)latency_tr / rate;
+ }
+
+ return (int)(latency + 0.5);
+ }
+
+ default :
+ return 0;
+ }
+}
+
+
+// Clears all the samples in the object's output and internal processing
+// buffers.
+void SoundTouch::clear()
+{
+ samplesExpectedOut = 0;
+ samplesOutput = 0;
+ pRateTransposer->clear();
+ pTDStretch->clear();
+}
+
+
+/// Returns number of samples currently unprocessed.
+uint SoundTouch::numUnprocessedSamples() const
+{
+ FIFOSamplePipe * psp;
+ if (pTDStretch)
+ {
+ psp = pTDStretch->getInput();
+ if (psp)
+ {
+ return psp->numSamples();
+ }
+ }
+ return 0;
+}
+
+
+/// Output samples from beginning of the sample buffer. Copies requested samples to
+/// output buffer and removes them from the sample buffer. If there are less than
+/// 'numsample' samples in the buffer, returns all that available.
+///
+/// \return Number of samples returned.
+uint SoundTouch::receiveSamples(SAMPLETYPE *output, uint maxSamples)
+{
+ uint ret = FIFOProcessor::receiveSamples(output, maxSamples);
+ samplesOutput += (long)ret;
+ return ret;
+}
+
+
+/// Adjusts book-keeping so that given number of samples are removed from beginning of the
+/// sample buffer without copying them anywhere.
+///
+/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+/// with 'ptrBegin' function.
+uint SoundTouch::receiveSamples(uint maxSamples)
+{
+ uint ret = FIFOProcessor::receiveSamples(maxSamples);
+ samplesOutput += (long)ret;
+ return ret;
+}
+
+
+/// Get ratio between input and output audio durations, useful for calculating
+/// processed output duration: if you'll process a stream of N samples, then
+/// you can expect to get out N * getInputOutputSampleRatio() samples.
+double SoundTouch::getInputOutputSampleRatio()
+{
+ return 1.0 / (tempo * rate);
+}
diff --git a/media/libsoundtouch/src/SoundTouch.h b/media/libsoundtouch/src/SoundTouch.h
new file mode 100644
index 0000000000..19c161fed5
--- /dev/null
+++ b/media/libsoundtouch/src/SoundTouch.h
@@ -0,0 +1,348 @@
+//////////////////////////////////////////////////////////////////////////////
+///
+/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
+///
+/// Notes:
+/// - Initialize the SoundTouch object instance by setting up the sound stream
+/// parameters with functions 'setSampleRate' and 'setChannels', then set
+/// desired tempo/pitch/rate settings with the corresponding functions.
+///
+/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
+/// samples that are to be processed are fed into one of the pipe by calling
+/// function 'putSamples', while the ready processed samples can be read
+/// from the other end of the pipeline with function 'receiveSamples'.
+///
+/// - The SoundTouch processing classes require certain sized 'batches' of
+/// samples in order to process the sound. For this reason the classes buffer
+/// incoming samples until there are enough of samples available for
+/// processing, then they carry out the processing step and consequently
+/// make the processed samples available for outputting.
+///
+/// - For the above reason, the processing routines introduce a certain
+/// 'latency' between the input and output, so that the samples input to
+/// SoundTouch may not be immediately available in the output, and neither
+/// the amount of outputtable samples may not immediately be in direct
+/// relationship with the amount of previously input samples.
+///
+/// - The tempo/pitch/rate control parameters can be altered during processing.
+/// Please notice though that they aren't currently protected by semaphores,
+/// so in multi-thread application external semaphore protection may be
+/// required.
+///
+/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
+/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
+/// tempo and pitch in the same ratio) of the sound. The third available control
+/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
+/// combining the two other controls.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef SoundTouch_H
+#define SoundTouch_H
+
+#include "FIFOSamplePipe.h"
+#include "STTypes.h"
+
+namespace soundtouch
+{
+
+/// Soundtouch library version string
+#define SOUNDTOUCH_VERSION "2.2"
+
+/// SoundTouch library version id
+#define SOUNDTOUCH_VERSION_ID (20200)
+
+//
+// Available setting IDs for the 'setSetting' & 'get_setting' functions:
+
+/// Enable/disable anti-alias filter in pitch transposer (0 = disable)
+#define SETTING_USE_AA_FILTER 0
+
+/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)
+#define SETTING_AA_FILTER_LENGTH 1
+
+/// Enable/disable quick seeking algorithm in tempo changer routine
+/// (enabling quick seeking lowers CPU utilization but causes a minor sound
+/// quality compromising)
+#define SETTING_USE_QUICKSEEK 2
+
+/// Time-stretch algorithm single processing sequence length in milliseconds. This determines
+/// to how long sequences the original sound is chopped in the time-stretch algorithm.
+/// See "STTypes.h" or README for more information.
+#define SETTING_SEQUENCE_MS 3
+
+/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the
+/// best possible overlapping location. This determines from how wide window the algorithm
+/// may look for an optimal joining location when mixing the sound sequences back together.
+/// See "STTypes.h" or README for more information.
+#define SETTING_SEEKWINDOW_MS 4
+
+/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences
+/// are mixed back together, to form a continuous sound stream, this parameter defines over
+/// how long period the two consecutive sequences are let to overlap each other.
+/// See "STTypes.h" or README for more information.
+#define SETTING_OVERLAP_MS 5
+
+
+/// Call "getSetting" with this ID to query processing sequence size in samples.
+/// This value gives approximate value of how many input samples you'll need to
+/// feed into SoundTouch after initial buffering to get out a new batch of
+/// output samples.
+///
+/// This value does not include initial buffering at beginning of a new processing
+/// stream, use SETTING_INITIAL_LATENCY to get the initial buffering size.
+///
+/// Notices:
+/// - This is read-only parameter, i.e. setSetting ignores this parameter
+/// - This parameter value is not constant but change depending on
+/// tempo/pitch/rate/samplerate settings.
+#define SETTING_NOMINAL_INPUT_SEQUENCE 6
+
+
+/// Call "getSetting" with this ID to query nominal average processing output
+/// size in samples. This value tells approcimate value how many output samples
+/// SoundTouch outputs once it does DSP processing run for a batch of input samples.
+///
+/// Notices:
+/// - This is read-only parameter, i.e. setSetting ignores this parameter
+/// - This parameter value is not constant but change depending on
+/// tempo/pitch/rate/samplerate settings.
+#define SETTING_NOMINAL_OUTPUT_SEQUENCE 7
+
+
+/// Call "getSetting" with this ID to query initial processing latency, i.e.
+/// approx. how many samples you'll need to enter to SoundTouch pipeline before
+/// you can expect to get first batch of ready output samples out.
+///
+/// After the first output batch, you can then expect to get approx.
+/// SETTING_NOMINAL_OUTPUT_SEQUENCE ready samples out for every
+/// SETTING_NOMINAL_INPUT_SEQUENCE samples that you enter into SoundTouch.
+///
+/// Example:
+/// processing with parameter -tempo=5
+/// => initial latency = 5509 samples
+/// input sequence = 4167 samples
+/// output sequence = 3969 samples
+///
+/// Accordingly, you can expect to feed in approx. 5509 samples at beginning of
+/// the stream, and then you'll get out the first 3969 samples. After that, for
+/// every approx. 4167 samples that you'll put in, you'll receive again approx.
+/// 3969 samples out.
+///
+/// This also means that average latency during stream processing is
+/// INITIAL_LATENCY-OUTPUT_SEQUENCE/2, in the above example case 5509-3969/2
+/// = 3524 samples
+///
+/// Notices:
+/// - This is read-only parameter, i.e. setSetting ignores this parameter
+/// - This parameter value is not constant but change depending on
+/// tempo/pitch/rate/samplerate settings.
+#define SETTING_INITIAL_LATENCY 8
+
+
+class SOUNDTOUCH_API SoundTouch : public FIFOProcessor
+{
+private:
+ /// Rate transposer class instance
+ class RateTransposer *pRateTransposer;
+
+ /// Time-stretch class instance
+ class TDStretch *pTDStretch;
+
+ /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
+ double virtualRate;
+
+ /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
+ double virtualTempo;
+
+ /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
+ double virtualPitch;
+
+ /// Flag: Has sample rate been set?
+ bool bSrateSet;
+
+ /// Accumulator for how many samples in total will be expected as output vs. samples put in,
+ /// considering current processing settings.
+ double samplesExpectedOut;
+
+ /// Accumulator for how many samples in total have been read out from the processing so far
+ long samplesOutput;
+
+ /// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
+ /// 'virtualPitch' parameters.
+ void calcEffectiveRateAndTempo();
+
+protected :
+ /// Number of channels
+ uint channels;
+
+ /// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
+ double rate;
+
+ /// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
+ double tempo;
+
+public:
+ SoundTouch();
+ virtual ~SoundTouch();
+
+ /// Get SoundTouch library version string
+ static const char *getVersionString();
+
+ /// Get SoundTouch library version Id
+ static uint getVersionId();
+
+ /// Sets new rate control value. Normal rate = 1.0, smaller values
+ /// represent slower rate, larger faster rates.
+ void setRate(double newRate);
+
+ /// Sets new tempo control value. Normal tempo = 1.0, smaller values
+ /// represent slower tempo, larger faster tempo.
+ void setTempo(double newTempo);
+
+ /// Sets new rate control value as a difference in percents compared
+ /// to the original rate (-50 .. +100 %)
+ void setRateChange(double newRate);
+
+ /// Sets new tempo control value as a difference in percents compared
+ /// to the original tempo (-50 .. +100 %)
+ void setTempoChange(double newTempo);
+
+ /// Sets new pitch control value. Original pitch = 1.0, smaller values
+ /// represent lower pitches, larger values higher pitch.
+ void setPitch(double newPitch);
+
+ /// Sets pitch change in octaves compared to the original pitch
+ /// (-1.00 .. +1.00)
+ void setPitchOctaves(double newPitch);
+
+ /// Sets pitch change in semi-tones compared to the original pitch
+ /// (-12 .. +12)
+ void setPitchSemiTones(int newPitch);
+ void setPitchSemiTones(double newPitch);
+
+ /// Sets the number of channels, 1 = mono, 2 = stereo
+ void setChannels(uint numChannels);
+
+ /// Sets sample rate.
+ void setSampleRate(uint srate);
+
+ /// Get ratio between input and output audio durations, useful for calculating
+ /// processed output duration: if you'll process a stream of N samples, then
+ /// you can expect to get out N * getInputOutputSampleRatio() samples.
+ ///
+ /// This ratio will give accurate target duration ratio for a full audio track,
+ /// given that the the whole track is processed with same processing parameters.
+ ///
+ /// If this ratio is applied to calculate intermediate offsets inside a processing
+ /// stream, then this ratio is approximate and can deviate +- some tens of milliseconds
+ /// from ideal offset, yet by end of the audio stream the duration ratio will become
+ /// exact.
+ ///
+ /// Example: if processing with parameters "-tempo=15 -pitch=-3", the function
+ /// will return value 0.8695652... Now, if processing an audio stream whose duration
+ /// is exactly one million audio samples, then you can expect the processed
+ /// output duration be 0.869565 * 1000000 = 869565 samples.
+ double getInputOutputSampleRatio();
+
+ /// Flushes the last samples from the processing pipeline to the output.
+ /// Clears also the internal processing buffers.
+ //
+ /// Note: This function is meant for extracting the last samples of a sound
+ /// stream. This function may introduce additional blank samples in the end
+ /// of the sound stream, and thus it's not recommended to call this function
+ /// in the middle of a sound stream.
+ void flush();
+
+ /// Adds 'numSamples' pcs of samples from the 'samples' memory position into
+ /// the input of the object. Notice that sample rate _has_to_ be set before
+ /// calling this function, otherwise throws a runtime_error exception.
+ virtual void putSamples(
+ const SAMPLETYPE *samples, ///< Pointer to sample buffer.
+ uint numSamples ///< Number of samples in buffer. Notice
+ ///< that in case of stereo-sound a single sample
+ ///< contains data for both channels.
+ );
+
+ /// Output samples from beginning of the sample buffer. Copies requested samples to
+ /// output buffer and removes them from the sample buffer. If there are less than
+ /// 'numsample' samples in the buffer, returns all that available.
+ ///
+ /// \return Number of samples returned.
+ virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
+ uint maxSamples ///< How many samples to receive at max.
+ );
+
+ /// Adjusts book-keeping so that given number of samples are removed from beginning of the
+ /// sample buffer without copying them anywhere.
+ ///
+ /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
+ /// with 'ptrBegin' function.
+ virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
+ );
+
+ /// Clears all the samples in the object's output and internal processing
+ /// buffers.
+ virtual void clear();
+
+ /// Changes a setting controlling the processing system behaviour. See the
+ /// 'SETTING_...' defines for available setting ID's.
+ ///
+ /// \return 'true' if the setting was successfully changed
+ bool setSetting(int settingId, ///< Setting ID number. see SETTING_... defines.
+ int value ///< New setting value.
+ );
+
+ /// Reads a setting controlling the processing system behaviour. See the
+ /// 'SETTING_...' defines for available setting ID's.
+ ///
+ /// \return the setting value.
+ int getSetting(int settingId ///< Setting ID number, see SETTING_... defines.
+ ) const;
+
+ /// Returns number of samples currently unprocessed.
+ virtual uint numUnprocessedSamples() const;
+
+ /// Return number of channels
+ uint numChannels() const
+ {
+ return channels;
+ }
+
+ /// Other handy functions that are implemented in the ancestor classes (see
+ /// classes 'FIFOProcessor' and 'FIFOSamplePipe')
+ ///
+ /// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch.
+ /// - numSamples() : Get number of 'ready' samples that can be received with
+ /// function 'receiveSamples()'
+ /// - isEmpty() : Returns nonzero if there aren't any 'ready' samples.
+ /// - clear() : Clears all samples from ready/processing buffers.
+};
+
+}
+#endif
diff --git a/media/libsoundtouch/src/SoundTouchFactory.cpp b/media/libsoundtouch/src/SoundTouchFactory.cpp
new file mode 100644
index 0000000000..d33fc133d7
--- /dev/null
+++ b/media/libsoundtouch/src/SoundTouchFactory.cpp
@@ -0,0 +1,31 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include <soundtouch/SoundTouch.h>
+
+namespace soundtouch
+{
+
+SOUNDTOUCH_API
+soundtouch::SoundTouch*
+createSoundTouchObj()
+{
+ return new soundtouch::SoundTouch();
+}
+
+SOUNDTOUCH_API
+void
+destroySoundTouchObj(soundtouch::SoundTouch* aObj)
+{
+ // SoundTouch runs deletes in its destructor, meaning they need to be run in
+ // the DLL context. Gecko should send its SoundTouch obj pointers here to be
+ // cleaned up.
+ if (aObj) {
+ delete aObj;
+ }
+}
+
+}
diff --git a/media/libsoundtouch/src/SoundTouchFactory.h b/media/libsoundtouch/src/SoundTouchFactory.h
new file mode 100644
index 0000000000..98d8a7deb1
--- /dev/null
+++ b/media/libsoundtouch/src/SoundTouchFactory.h
@@ -0,0 +1,22 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Code for dealing with creating/deleting SoundTouch objects across DLL
+// boundaries.
+
+#include <soundtouch/STTypes.h>
+#include <soundtouch/SoundTouch.h>
+
+namespace soundtouch
+{
+SOUNDTOUCH_API
+soundtouch::SoundTouch*
+createSoundTouchObj();
+
+SOUNDTOUCH_API
+void
+destroySoundTouchObj(soundtouch::SoundTouch* aObj);
+}
diff --git a/media/libsoundtouch/src/TDStretch.cpp b/media/libsoundtouch/src/TDStretch.cpp
new file mode 100644
index 0000000000..709e979d1d
--- /dev/null
+++ b/media/libsoundtouch/src/TDStretch.cpp
@@ -0,0 +1,1108 @@
+///////////////////////////////////////////////////////////////////////////////
+///
+/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
+/// while maintaining the original pitch by using a time domain WSOLA-like
+/// method with several performance-increasing tweaks.
+///
+/// Notes : MMX optimized functions reside in a separate, platform-specific
+/// file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'.
+///
+/// This source file contains OpenMP optimizations that allow speeding up the
+/// corss-correlation algorithm by executing it in several threads / CPU cores
+/// in parallel. See the following article link for more detailed discussion
+/// about SoundTouch OpenMP optimizations:
+/// http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include <string.h>
+#include <limits.h>
+#include <assert.h>
+#include <math.h>
+#include <float.h>
+
+#include "STTypes.h"
+#include "cpu_detect.h"
+#include "TDStretch.h"
+
+using namespace soundtouch;
+
+#define max(x, y) (((x) > (y)) ? (x) : (y))
+
+/*****************************************************************************
+ *
+ * Constant definitions
+ *
+ *****************************************************************************/
+
+// Table for the hierarchical mixing position seeking algorithm
+const short _scanOffsets[5][24]={
+ { 124, 186, 248, 310, 372, 434, 496, 558, 620, 682, 744, 806,
+ 868, 930, 992, 1054, 1116, 1178, 1240, 1302, 1364, 1426, 1488, 0},
+ {-100, -75, -50, -25, 25, 50, 75, 100, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
+ { -20, -15, -10, -5, 5, 10, 15, 20, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
+ { -4, -3, -2, -1, 1, 2, 3, 4, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
+ { 121, 114, 97, 114, 98, 105, 108, 32, 104, 99, 117, 111,
+ 116, 100, 110, 117, 111, 115, 0, 0, 0, 0, 0, 0}};
+
+/*****************************************************************************
+ *
+ * Implementation of the class 'TDStretch'
+ *
+ *****************************************************************************/
+
+
+TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
+{
+ bQuickSeek = false;
+ channels = 2;
+
+ pMidBuffer = NULL;
+ pMidBufferUnaligned = NULL;
+ overlapLength = 0;
+
+ bAutoSeqSetting = true;
+ bAutoSeekSetting = true;
+
+ tempo = 1.0f;
+ setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS);
+ setTempo(1.0f);
+
+ clear();
+}
+
+
+
+TDStretch::~TDStretch()
+{
+ delete[] pMidBufferUnaligned;
+}
+
+
+
+// Sets routine control parameters. These control are certain time constants
+// defining how the sound is stretched to the desired duration.
+//
+// 'sampleRate' = sample rate of the sound
+// 'sequenceMS' = one processing sequence length in milliseconds (default = 82 ms)
+// 'seekwindowMS' = seeking window length for scanning the best overlapping
+// position (default = 28 ms)
+// 'overlapMS' = overlapping length (default = 12 ms)
+
+void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
+ int aSeekWindowMS, int aOverlapMS)
+{
+ // accept only positive parameter values - if zero or negative, use old values instead
+ if (aSampleRate > 0)
+ {
+ if (aSampleRate > 192000) ST_THROW_RT_ERROR("Error: Excessive samplerate");
+ this->sampleRate = aSampleRate;
+ }
+
+ if (aOverlapMS > 0) this->overlapMs = aOverlapMS;
+
+ if (aSequenceMS > 0)
+ {
+ this->sequenceMs = aSequenceMS;
+ bAutoSeqSetting = false;
+ }
+ else if (aSequenceMS == 0)
+ {
+ // if zero, use automatic setting
+ bAutoSeqSetting = true;
+ }
+
+ if (aSeekWindowMS > 0)
+ {
+ this->seekWindowMs = aSeekWindowMS;
+ bAutoSeekSetting = false;
+ }
+ else if (aSeekWindowMS == 0)
+ {
+ // if zero, use automatic setting
+ bAutoSeekSetting = true;
+ }
+
+ calcSeqParameters();
+
+ calculateOverlapLength(overlapMs);
+
+ // set tempo to recalculate 'sampleReq'
+ setTempo(tempo);
+}
+
+
+
+/// Get routine control parameters, see setParameters() function.
+/// Any of the parameters to this function can be NULL, in such case corresponding parameter
+/// value isn't returned.
+void TDStretch::getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const
+{
+ if (pSampleRate)
+ {
+ *pSampleRate = sampleRate;
+ }
+
+ if (pSequenceMs)
+ {
+ *pSequenceMs = (bAutoSeqSetting) ? (USE_AUTO_SEQUENCE_LEN) : sequenceMs;
+ }
+
+ if (pSeekWindowMs)
+ {
+ *pSeekWindowMs = (bAutoSeekSetting) ? (USE_AUTO_SEEKWINDOW_LEN) : seekWindowMs;
+ }
+
+ if (pOverlapMs)
+ {
+ *pOverlapMs = overlapMs;
+ }
+}
+
+
+// Overlaps samples in 'midBuffer' with the samples in 'pInput'
+void TDStretch::overlapMono(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput) const
+{
+ int i;
+ SAMPLETYPE m1, m2;
+
+ m1 = (SAMPLETYPE)0;
+ m2 = (SAMPLETYPE)overlapLength;
+
+ for (i = 0; i < overlapLength ; i ++)
+ {
+ pOutput[i] = (pInput[i] * m1 + pMidBuffer[i] * m2 ) / overlapLength;
+ m1 += 1;
+ m2 -= 1;
+ }
+}
+
+
+
+void TDStretch::clearMidBuffer()
+{
+ memset(pMidBuffer, 0, channels * sizeof(SAMPLETYPE) * overlapLength);
+}
+
+
+void TDStretch::clearInput()
+{
+ inputBuffer.clear();
+ clearMidBuffer();
+ isBeginning = true;
+ maxnorm = 0;
+ maxnormf = 1e8;
+ skipFract = 0;
+}
+
+
+// Clears the sample buffers
+void TDStretch::clear()
+{
+ outputBuffer.clear();
+ clearInput();
+}
+
+
+
+// Enables/disables the quick position seeking algorithm. Zero to disable, nonzero
+// to enable
+void TDStretch::enableQuickSeek(bool enable)
+{
+ bQuickSeek = enable;
+}
+
+
+// Returns nonzero if the quick seeking algorithm is enabled.
+bool TDStretch::isQuickSeekEnabled() const
+{
+ return bQuickSeek;
+}
+
+
+// Seeks for the optimal overlap-mixing position.
+int TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos)
+{
+ if (bQuickSeek)
+ {
+ return seekBestOverlapPositionQuick(refPos);
+ }
+ else
+ {
+ return seekBestOverlapPositionFull(refPos);
+ }
+}
+
+
+// Overlaps samples in 'midBuffer' with the samples in 'pInputBuffer' at position
+// of 'ovlPos'.
+inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, uint ovlPos) const
+{
+#ifndef USE_MULTICH_ALWAYS
+ if (channels == 1)
+ {
+ // mono sound.
+ overlapMono(pOutput, pInput + ovlPos);
+ }
+ else if (channels == 2)
+ {
+ // stereo sound
+ overlapStereo(pOutput, pInput + 2 * ovlPos);
+ }
+ else
+#endif // USE_MULTICH_ALWAYS
+ {
+ assert(channels > 0);
+ overlapMulti(pOutput, pInput + channels * ovlPos);
+ }
+}
+
+
+// Seeks for the optimal overlap-mixing position. The 'stereo' version of the
+// routine
+//
+// The best position is determined as the position where the two overlapped
+// sample sequences are 'most alike', in terms of the highest cross-correlation
+// value over the overlapping period
+int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
+{
+ int bestOffs;
+ double bestCorr;
+ int i;
+ double norm;
+
+ bestCorr = -FLT_MAX;
+ bestOffs = 0;
+
+ // Scans for the best correlation value by testing each possible position
+ // over the permitted range.
+ bestCorr = calcCrossCorr(refPos, pMidBuffer, norm);
+ bestCorr = (bestCorr + 0.1) * 0.75;
+
+ #pragma omp parallel for
+ for (i = 1; i < seekLength; i ++)
+ {
+ double corr;
+ // Calculates correlation value for the mixing position corresponding to 'i'
+#if defined(_OPENMP) || defined(ST_SIMD_AVOID_UNALIGNED)
+ // in parallel OpenMP mode, can't use norm accumulator version as parallel executor won't
+ // iterate the loop in sequential order
+ // in SIMD mode, avoid accumulator version to allow avoiding unaligned positions
+ corr = calcCrossCorr(refPos + channels * i, pMidBuffer, norm);
+#else
+ // In non-parallel version call "calcCrossCorrAccumulate" that is otherwise same
+ // as "calcCrossCorr", but saves time by reusing & updating previously stored
+ // "norm" value
+ corr = calcCrossCorrAccumulate(refPos + channels * i, pMidBuffer, norm);
+#endif
+ // heuristic rule to slightly favour values close to mid of the range
+ double tmp = (double)(2 * i - seekLength) / (double)seekLength;
+ corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));
+
+ // Checks for the highest correlation value
+ if (corr > bestCorr)
+ {
+ // For optimal performance, enter critical section only in case that best value found.
+ // in such case repeat 'if' condition as it's possible that parallel execution may have
+ // updated the bestCorr value in the mean time
+ #pragma omp critical
+ if (corr > bestCorr)
+ {
+ bestCorr = corr;
+ bestOffs = i;
+ }
+ }
+ }
+
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+ adaptNormalizer();
+#endif
+
+ // clear cross correlation routine state if necessary (is so e.g. in MMX routines).
+ clearCrossCorrState();
+
+ return bestOffs;
+}
+
+
+// Quick seek algorithm for improved runtime-performance: First roughly scans through the
+// correlation area, and then scan surroundings of two best preliminary correlation candidates
+// with improved precision
+//
+// Based on testing:
+// - This algorithm gives on average 99% as good match as the full algorithm
+// - this quick seek algorithm finds the best match on ~90% of cases
+// - on those 10% of cases when this algorithm doesn't find best match,
+// it still finds on average ~90% match vs. the best possible match
+int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos)
+{
+#define _MIN(a, b) (((a) < (b)) ? (a) : (b))
+#define SCANSTEP 16
+#define SCANWIND 8
+
+ int bestOffs;
+ int i;
+ int bestOffs2;
+ float bestCorr, corr;
+ float bestCorr2;
+ double norm;
+
+ // note: 'float' types used in this function in case that the platform would need to use software-fp
+
+ bestCorr =
+ bestCorr2 = -FLT_MAX;
+ bestOffs =
+ bestOffs2 = SCANWIND;
+
+ // Scans for the best correlation value by testing each possible position
+ // over the permitted range. Look for two best matches on the first pass to
+ // increase possibility of ideal match.
+ //
+ // Begin from "SCANSTEP" instead of SCANWIND to make the calculation
+ // catch the 'middlepoint' of seekLength vector as that's the a-priori
+ // expected best match position
+ //
+ // Roughly:
+ // - 15% of cases find best result directly on the first round,
+ // - 75% cases find better match on 2nd round around the best match from 1st round
+ // - 10% cases find better match on 2nd round around the 2nd-best-match from 1st round
+ for (i = SCANSTEP; i < seekLength - SCANWIND - 1; i += SCANSTEP)
+ {
+ // Calculates correlation value for the mixing position corresponding
+ // to 'i'
+ corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm);
+ // heuristic rule to slightly favour values close to mid of the seek range
+ float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength;
+ corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp));
+
+ // Checks for the highest correlation value
+ if (corr > bestCorr)
+ {
+ // found new best match. keep the previous best as 2nd best match
+ bestCorr2 = bestCorr;
+ bestOffs2 = bestOffs;
+ bestCorr = corr;
+ bestOffs = i;
+ }
+ else if (corr > bestCorr2)
+ {
+ // not new best, but still new 2nd best match
+ bestCorr2 = corr;
+ bestOffs2 = i;
+ }
+ }
+
+ // Scans surroundings of the found best match with small stepping
+ int end = _MIN(bestOffs + SCANWIND + 1, seekLength);
+ for (i = bestOffs - SCANWIND; i < end; i++)
+ {
+ if (i == bestOffs) continue; // this offset already calculated, thus skip
+
+ // Calculates correlation value for the mixing position corresponding
+ // to 'i'
+ corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm);
+ // heuristic rule to slightly favour values close to mid of the range
+ float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength;
+ corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp));
+
+ // Checks for the highest correlation value
+ if (corr > bestCorr)
+ {
+ bestCorr = corr;
+ bestOffs = i;
+ }
+ }
+
+ // Scans surroundings of the 2nd best match with small stepping
+ end = _MIN(bestOffs2 + SCANWIND + 1, seekLength);
+ for (i = bestOffs2 - SCANWIND; i < end; i++)
+ {
+ if (i == bestOffs2) continue; // this offset already calculated, thus skip
+
+ // Calculates correlation value for the mixing position corresponding
+ // to 'i'
+ corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm);
+ // heuristic rule to slightly favour values close to mid of the range
+ float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength;
+ corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp));
+
+ // Checks for the highest correlation value
+ if (corr > bestCorr)
+ {
+ bestCorr = corr;
+ bestOffs = i;
+ }
+ }
+
+ // clear cross correlation routine state if necessary (is so e.g. in MMX routines).
+ clearCrossCorrState();
+
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+ adaptNormalizer();
+#endif
+
+ return bestOffs;
+}
+
+
+
+
+/// For integer algorithm: adapt normalization factor divider with music so that
+/// it'll not be pessimistically restrictive that can degrade quality on quieter sections
+/// yet won't cause integer overflows either
+void TDStretch::adaptNormalizer()
+{
+ // Do not adapt normalizer over too silent sequences to avoid averaging filter depleting to
+ // too low values during pauses in music
+ if ((maxnorm > 1000) || (maxnormf > 40000000))
+ {
+ //norm averaging filter
+ maxnormf = 0.9f * maxnormf + 0.1f * (float)maxnorm;
+
+ if ((maxnorm > 800000000) && (overlapDividerBitsNorm < 16))
+ {
+ // large values, so increase divider
+ overlapDividerBitsNorm++;
+ if (maxnorm > 1600000000) overlapDividerBitsNorm++; // extra large value => extra increase
+ }
+ else if ((maxnormf < 1000000) && (overlapDividerBitsNorm > 0))
+ {
+ // extra small values, decrease divider
+ overlapDividerBitsNorm--;
+ }
+ }
+
+ maxnorm = 0;
+}
+
+
+/// clear cross correlation routine state if necessary
+void TDStretch::clearCrossCorrState()
+{
+ // default implementation is empty.
+}
+
+
+/// Calculates processing sequence length according to tempo setting
+void TDStretch::calcSeqParameters()
+{
+ // Adjust tempo param according to tempo, so that variating processing sequence length is used
+ // at various tempo settings, between the given low...top limits
+ #define AUTOSEQ_TEMPO_LOW 0.5 // auto setting low tempo range (-50%)
+ #define AUTOSEQ_TEMPO_TOP 2.0 // auto setting top tempo range (+100%)
+
+ // sequence-ms setting values at above low & top tempo
+ #define AUTOSEQ_AT_MIN 90.0
+ #define AUTOSEQ_AT_MAX 40.0
+ #define AUTOSEQ_K ((AUTOSEQ_AT_MAX - AUTOSEQ_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))
+ #define AUTOSEQ_C (AUTOSEQ_AT_MIN - (AUTOSEQ_K) * (AUTOSEQ_TEMPO_LOW))
+
+ // seek-window-ms setting values at above low & top tempoq
+ #define AUTOSEEK_AT_MIN 20.0
+ #define AUTOSEEK_AT_MAX 15.0
+ #define AUTOSEEK_K ((AUTOSEEK_AT_MAX - AUTOSEEK_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))
+ #define AUTOSEEK_C (AUTOSEEK_AT_MIN - (AUTOSEEK_K) * (AUTOSEQ_TEMPO_LOW))
+
+ #define CHECK_LIMITS(x, mi, ma) (((x) < (mi)) ? (mi) : (((x) > (ma)) ? (ma) : (x)))
+
+ double seq, seek;
+
+ if (bAutoSeqSetting)
+ {
+ seq = AUTOSEQ_C + AUTOSEQ_K * tempo;
+ seq = CHECK_LIMITS(seq, AUTOSEQ_AT_MAX, AUTOSEQ_AT_MIN);
+ sequenceMs = (int)(seq + 0.5);
+ }
+
+ if (bAutoSeekSetting)
+ {
+ seek = AUTOSEEK_C + AUTOSEEK_K * tempo;
+ seek = CHECK_LIMITS(seek, AUTOSEEK_AT_MAX, AUTOSEEK_AT_MIN);
+ seekWindowMs = (int)(seek + 0.5);
+ }
+
+ // Update seek window lengths
+ seekWindowLength = (sampleRate * sequenceMs) / 1000;
+ if (seekWindowLength < 2 * overlapLength)
+ {
+ seekWindowLength = 2 * overlapLength;
+ }
+ seekLength = (sampleRate * seekWindowMs) / 1000;
+}
+
+
+
+// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
+// tempo, larger faster tempo.
+void TDStretch::setTempo(double newTempo)
+{
+ int intskip;
+
+ tempo = newTempo;
+
+ // Calculate new sequence duration
+ calcSeqParameters();
+
+ // Calculate ideal skip length (according to tempo value)
+ nominalSkip = tempo * (seekWindowLength - overlapLength);
+ intskip = (int)(nominalSkip + 0.5);
+
+ // Calculate how many samples are needed in the 'inputBuffer' to
+ // process another batch of samples
+ //sampleReq = max(intskip + overlapLength, seekWindowLength) + seekLength / 2;
+ sampleReq = max(intskip + overlapLength, seekWindowLength) + seekLength;
+}
+
+
+
+// Sets the number of channels, 1 = mono, 2 = stereo
+void TDStretch::setChannels(int numChannels)
+{
+ if (!verifyNumberOfChannels(numChannels) ||
+ (channels == numChannels)) return;
+
+ channels = numChannels;
+ inputBuffer.setChannels(channels);
+ outputBuffer.setChannels(channels);
+
+ // re-init overlap/buffer
+ overlapLength=0;
+ setParameters(sampleRate);
+}
+
+
+// nominal tempo, no need for processing, just pass the samples through
+// to outputBuffer
+/*
+void TDStretch::processNominalTempo()
+{
+ assert(tempo == 1.0f);
+
+ if (bMidBufferDirty)
+ {
+ // If there are samples in pMidBuffer waiting for overlapping,
+ // do a single sliding overlapping with them in order to prevent a
+ // clicking distortion in the output sound
+ if (inputBuffer.numSamples() < overlapLength)
+ {
+ // wait until we've got overlapLength input samples
+ return;
+ }
+ // Mix the samples in the beginning of 'inputBuffer' with the
+ // samples in 'midBuffer' using sliding overlapping
+ overlap(outputBuffer.ptrEnd(overlapLength), inputBuffer.ptrBegin(), 0);
+ outputBuffer.putSamples(overlapLength);
+ inputBuffer.receiveSamples(overlapLength);
+ clearMidBuffer();
+ // now we've caught the nominal sample flow and may switch to
+ // bypass mode
+ }
+
+ // Simply bypass samples from input to output
+ outputBuffer.moveSamples(inputBuffer);
+}
+*/
+
+
+// Processes as many processing frames of the samples 'inputBuffer', store
+// the result into 'outputBuffer'
+void TDStretch::processSamples()
+{
+ int ovlSkip;
+ int offset = 0;
+ int temp;
+
+ /* Removed this small optimization - can introduce a click to sound when tempo setting
+ crosses the nominal value
+ if (tempo == 1.0f)
+ {
+ // tempo not changed from the original, so bypass the processing
+ processNominalTempo();
+ return;
+ }
+ */
+
+ // Process samples as long as there are enough samples in 'inputBuffer'
+ // to form a processing frame.
+ while ((int)inputBuffer.numSamples() >= sampleReq)
+ {
+ if (isBeginning == false)
+ {
+ // apart from the very beginning of the track,
+ // scan for the best overlapping position & do overlap-add
+ offset = seekBestOverlapPosition(inputBuffer.ptrBegin());
+
+ // Mix the samples in the 'inputBuffer' at position of 'offset' with the
+ // samples in 'midBuffer' using sliding overlapping
+ // ... first partially overlap with the end of the previous sequence
+ // (that's in 'midBuffer')
+ overlap(outputBuffer.ptrEnd((uint)overlapLength), inputBuffer.ptrBegin(), (uint)offset);
+ outputBuffer.putSamples((uint)overlapLength);
+ offset += overlapLength;
+ }
+ else
+ {
+ // Adjust processing offset at beginning of track by not perform initial overlapping
+ // and compensating that in the 'input buffer skip' calculation
+ isBeginning = false;
+ int skip = (int)(tempo * overlapLength + 0.5 * seekLength + 0.5);
+
+ #ifdef ST_SIMD_AVOID_UNALIGNED
+ // in SIMD mode, round the skip amount to value corresponding to aligned memory address
+ if (channels == 1)
+ {
+ skip &= -4;
+ }
+ else if (channels == 2)
+ {
+ skip &= -2;
+ }
+ #endif
+ skipFract -= skip;
+ if (skipFract <= -nominalSkip)
+ {
+ skipFract = -nominalSkip;
+ }
+ }
+
+ // ... then copy sequence samples from 'inputBuffer' to output:
+
+ // crosscheck that we don't have buffer overflow...
+ if ((int)inputBuffer.numSamples() < (offset + seekWindowLength - overlapLength))
+ {
+ continue; // just in case, shouldn't really happen
+ }
+
+ // length of sequence
+ temp = (seekWindowLength - 2 * overlapLength);
+ outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * offset, (uint)temp);
+
+ // Copies the end of the current sequence from 'inputBuffer' to
+ // 'midBuffer' for being mixed with the beginning of the next
+ // processing sequence and so on
+ assert((offset + temp + overlapLength) <= (int)inputBuffer.numSamples());
+ memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + temp),
+ channels * sizeof(SAMPLETYPE) * overlapLength);
+
+ // Remove the processed samples from the input buffer. Update
+ // the difference between integer & nominal skip step to 'skipFract'
+ // in order to prevent the error from accumulating over time.
+ skipFract += nominalSkip; // real skip size
+ ovlSkip = (int)skipFract; // rounded to integer skip
+ skipFract -= ovlSkip; // maintain the fraction part, i.e. real vs. integer skip
+ inputBuffer.receiveSamples((uint)ovlSkip);
+ }
+}
+
+
+// Adds 'numsamples' pcs of samples from the 'samples' memory position into
+// the input of the object.
+void TDStretch::putSamples(const SAMPLETYPE *samples, uint nSamples)
+{
+ // Add the samples into the input buffer
+ inputBuffer.putSamples(samples, nSamples);
+ // Process the samples in input buffer
+ processSamples();
+}
+
+
+
+/// Set new overlap length parameter & reallocate RefMidBuffer if necessary.
+void TDStretch::acceptNewOverlapLength(int newOverlapLength)
+{
+ int prevOvl;
+
+ assert(newOverlapLength >= 0);
+ prevOvl = overlapLength;
+ overlapLength = newOverlapLength;
+
+ if (overlapLength > prevOvl)
+ {
+ delete[] pMidBufferUnaligned;
+
+ pMidBufferUnaligned = new SAMPLETYPE[overlapLength * channels + 16 / sizeof(SAMPLETYPE)];
+ // ensure that 'pMidBuffer' is aligned to 16 byte boundary for efficiency
+ pMidBuffer = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(pMidBufferUnaligned);
+
+ clearMidBuffer();
+ }
+}
+
+
+// Operator 'new' is overloaded so that it automatically creates a suitable instance
+// depending on if we've a MMX/SSE/etc-capable CPU available or not.
+void * TDStretch::operator new(size_t s)
+{
+ // Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!
+ ST_THROW_RT_ERROR("Error in TDStretch::new: Don't use 'new TDStretch' directly, use 'newInstance' member instead!");
+ return newInstance();
+}
+
+
+TDStretch * TDStretch::newInstance()
+{
+#if defined(SOUNDTOUCH_ALLOW_MMX) || defined(SOUNDTOUCH_ALLOW_SSE)
+ uint uExtensions;
+
+ uExtensions = detectCPUextensions();
+#endif
+
+ // Check if MMX/SSE instruction set extensions supported by CPU
+
+#ifdef SOUNDTOUCH_ALLOW_MMX
+ // MMX routines available only with integer sample types
+ if (uExtensions & SUPPORT_MMX)
+ {
+ return ::new TDStretchMMX;
+ }
+ else
+#endif // SOUNDTOUCH_ALLOW_MMX
+
+
+#ifdef SOUNDTOUCH_ALLOW_SSE
+ if (uExtensions & SUPPORT_SSE)
+ {
+ // SSE support
+ return ::new TDStretchSSE;
+ }
+ else
+#endif // SOUNDTOUCH_ALLOW_SSE
+
+ {
+ // ISA optimizations not supported, use plain C version
+ return ::new TDStretch;
+ }
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// Integer arithmetic specific algorithm implementations.
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+
+// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Stereo'
+// version of the routine.
+void TDStretch::overlapStereo(short *poutput, const short *input) const
+{
+ int i;
+ short temp;
+ int cnt2;
+
+ for (i = 0; i < overlapLength ; i ++)
+ {
+ temp = (short)(overlapLength - i);
+ cnt2 = 2 * i;
+ poutput[cnt2] = (input[cnt2] * i + pMidBuffer[cnt2] * temp ) / overlapLength;
+ poutput[cnt2 + 1] = (input[cnt2 + 1] * i + pMidBuffer[cnt2 + 1] * temp ) / overlapLength;
+ }
+}
+
+
+// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Multi'
+// version of the routine.
+void TDStretch::overlapMulti(short *poutput, const short *input) const
+{
+ short m1;
+ int i = 0;
+
+ for (m1 = 0; m1 < overlapLength; m1 ++)
+ {
+ short m2 = (short)(overlapLength - m1);
+ for (int c = 0; c < channels; c ++)
+ {
+ poutput[i] = (input[i] * m1 + pMidBuffer[i] * m2) / overlapLength;
+ i++;
+ }
+ }
+}
+
+// Calculates the x having the closest 2^x value for the given value
+static int _getClosest2Power(double value)
+{
+ return (int)(log(value) / log(2.0) + 0.5);
+}
+
+
+/// Calculates overlap period length in samples.
+/// Integer version rounds overlap length to closest power of 2
+/// for a divide scaling operation.
+void TDStretch::calculateOverlapLength(int aoverlapMs)
+{
+ int newOvl;
+
+ assert(aoverlapMs >= 0);
+
+ // calculate overlap length so that it's power of 2 - thus it's easy to do
+ // integer division by right-shifting. Term "-1" at end is to account for
+ // the extra most significatnt bit left unused in result by signed multiplication
+ overlapDividerBitsPure = _getClosest2Power((sampleRate * aoverlapMs) / 1000.0) - 1;
+ if (overlapDividerBitsPure > 9) overlapDividerBitsPure = 9;
+ if (overlapDividerBitsPure < 3) overlapDividerBitsPure = 3;
+ newOvl = (int)pow(2.0, (int)overlapDividerBitsPure + 1); // +1 => account for -1 above
+
+ acceptNewOverlapLength(newOvl);
+
+ overlapDividerBitsNorm = overlapDividerBitsPure;
+
+ // calculate sloping divider so that crosscorrelation operation won't
+ // overflow 32-bit register. Max. sum of the crosscorrelation sum without
+ // divider would be 2^30*(N^3-N)/3, where N = overlap length
+ slopingDivider = (newOvl * newOvl - 1) / 3;
+}
+
+
+double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, double &norm)
+{
+ long corr;
+ unsigned long lnorm;
+ int i;
+
+ #ifdef ST_SIMD_AVOID_UNALIGNED
+ // in SIMD mode skip 'mixingPos' positions that aren't aligned to 16-byte boundary
+ if (((ulongptr)mixingPos) & 15) return -1e50;
+ #endif
+
+ // hint compiler autovectorization that loop length is divisible by 8
+ int ilength = (channels * overlapLength) & -8;
+
+ corr = lnorm = 0;
+ // Same routine for stereo and mono
+ for (i = 0; i < ilength; i += 2)
+ {
+ corr += (mixingPos[i] * compare[i] +
+ mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm;
+ lnorm += (mixingPos[i] * mixingPos[i] +
+ mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBitsNorm;
+ // do intermediate scalings to avoid integer overflow
+ }
+
+ if (lnorm > maxnorm)
+ {
+ // modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode
+ #pragma omp critical
+ if (lnorm > maxnorm)
+ {
+ maxnorm = lnorm;
+ }
+ }
+ // Normalize result by dividing by sqrt(norm) - this step is easiest
+ // done using floating point operation
+ norm = (double)lnorm;
+ return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm);
+}
+
+
+/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
+double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm)
+{
+ long corr;
+ long lnorm;
+ int i;
+
+ // hint compiler autovectorization that loop length is divisible by 8
+ int ilength = (channels * overlapLength) & -8;
+
+ // cancel first normalizer tap from previous round
+ lnorm = 0;
+ for (i = 1; i <= channels; i ++)
+ {
+ lnorm -= (mixingPos[-i] * mixingPos[-i]) >> overlapDividerBitsNorm;
+ }
+
+ corr = 0;
+ // Same routine for stereo and mono.
+ for (i = 0; i < ilength; i += 2)
+ {
+ corr += (mixingPos[i] * compare[i] +
+ mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm;
+ }
+
+ // update normalizer with last samples of this round
+ for (int j = 0; j < channels; j ++)
+ {
+ i --;
+ lnorm += (mixingPos[i] * mixingPos[i]) >> overlapDividerBitsNorm;
+ }
+
+ norm += (double)lnorm;
+ if (norm > maxnorm)
+ {
+ maxnorm = (unsigned long)norm;
+ }
+
+ // Normalize result by dividing by sqrt(norm) - this step is easiest
+ // done using floating point operation
+ return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm);
+}
+
+#endif // SOUNDTOUCH_INTEGER_SAMPLES
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// Floating point arithmetic specific algorithm implementations.
+//
+
+#ifdef SOUNDTOUCH_FLOAT_SAMPLES
+
+// Overlaps samples in 'midBuffer' with the samples in 'pInput'
+void TDStretch::overlapStereo(float *pOutput, const float *pInput) const
+{
+ int i;
+ float fScale;
+ float f1;
+ float f2;
+
+ fScale = 1.0f / (float)overlapLength;
+
+ f1 = 0;
+ f2 = 1.0f;
+
+ for (i = 0; i < 2 * (int)overlapLength ; i += 2)
+ {
+ pOutput[i + 0] = pInput[i + 0] * f1 + pMidBuffer[i + 0] * f2;
+ pOutput[i + 1] = pInput[i + 1] * f1 + pMidBuffer[i + 1] * f2;
+
+ f1 += fScale;
+ f2 -= fScale;
+ }
+}
+
+
+// Overlaps samples in 'midBuffer' with the samples in 'input'.
+void TDStretch::overlapMulti(float *pOutput, const float *pInput) const
+{
+ int i;
+ float fScale;
+ float f1;
+ float f2;
+
+ fScale = 1.0f / (float)overlapLength;
+
+ f1 = 0;
+ f2 = 1.0f;
+
+ i=0;
+ for (int i2 = 0; i2 < overlapLength; i2 ++)
+ {
+ // note: Could optimize this slightly by taking into account that always channels > 2
+ for (int c = 0; c < channels; c ++)
+ {
+ pOutput[i] = pInput[i] * f1 + pMidBuffer[i] * f2;
+ i++;
+ }
+ f1 += fScale;
+ f2 -= fScale;
+ }
+}
+
+
+/// Calculates overlapInMsec period length in samples.
+void TDStretch::calculateOverlapLength(int overlapInMsec)
+{
+ int newOvl;
+
+ assert(overlapInMsec >= 0);
+ newOvl = (sampleRate * overlapInMsec) / 1000;
+ if (newOvl < 16) newOvl = 16;
+
+ // must be divisible by 8
+ newOvl -= newOvl % 8;
+
+ acceptNewOverlapLength(newOvl);
+}
+
+
+/// Calculate cross-correlation
+double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, double &anorm)
+{
+ float corr;
+ float norm;
+ int i;
+
+ #ifdef ST_SIMD_AVOID_UNALIGNED
+ // in SIMD mode skip 'mixingPos' positions that aren't aligned to 16-byte boundary
+ if (((ulongptr)mixingPos) & 15) return -1e50;
+ #endif
+
+ // hint compiler autovectorization that loop length is divisible by 8
+ int ilength = (channels * overlapLength) & -8;
+
+ corr = norm = 0;
+ // Same routine for stereo and mono
+ for (i = 0; i < ilength; i ++)
+ {
+ corr += mixingPos[i] * compare[i];
+ norm += mixingPos[i] * mixingPos[i];
+ }
+
+ anorm = norm;
+ return corr / sqrt((norm < 1e-9 ? 1.0 : norm));
+}
+
+
+/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
+double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm)
+{
+ float corr;
+ int i;
+
+ corr = 0;
+
+ // cancel first normalizer tap from previous round
+ for (i = 1; i <= channels; i ++)
+ {
+ norm -= mixingPos[-i] * mixingPos[-i];
+ }
+
+ // hint compiler autovectorization that loop length is divisible by 8
+ int ilength = (channels * overlapLength) & -8;
+
+ // Same routine for stereo and mono
+ for (i = 0; i < ilength; i ++)
+ {
+ corr += mixingPos[i] * compare[i];
+ }
+
+ // update normalizer with last samples of this round
+ for (int j = 0; j < channels; j ++)
+ {
+ i --;
+ norm += mixingPos[i] * mixingPos[i];
+ }
+
+ return corr / sqrt((norm < 1e-9 ? 1.0 : norm));
+}
+
+
+#endif // SOUNDTOUCH_FLOAT_SAMPLES
diff --git a/media/libsoundtouch/src/TDStretch.h b/media/libsoundtouch/src/TDStretch.h
new file mode 100644
index 0000000000..3ef79c7cdf
--- /dev/null
+++ b/media/libsoundtouch/src/TDStretch.h
@@ -0,0 +1,279 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
+/// while maintaining the original pitch by using a time domain WSOLA-like method
+/// with several performance-increasing tweaks.
+///
+/// Note : MMX/SSE optimized functions reside in separate, platform-specific files
+/// 'mmx_optimized.cpp' and 'sse_optimized.cpp'
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef TDStretch_H
+#define TDStretch_H
+
+#include <stddef.h>
+#include "STTypes.h"
+#include "RateTransposer.h"
+#include "FIFOSamplePipe.h"
+
+namespace soundtouch
+{
+
+/// Default values for sound processing parameters:
+/// Notice that the default parameters are tuned for contemporary popular music
+/// processing. For speech processing applications these parameters suit better:
+/// #define DEFAULT_SEQUENCE_MS 40
+/// #define DEFAULT_SEEKWINDOW_MS 15
+/// #define DEFAULT_OVERLAP_MS 8
+///
+
+/// Default length of a single processing sequence, in milliseconds. This determines to how
+/// long sequences the original sound is chopped in the time-stretch algorithm.
+///
+/// The larger this value is, the lesser sequences are used in processing. In principle
+/// a bigger value sounds better when slowing down tempo, but worse when increasing tempo
+/// and vice versa.
+///
+/// Increasing this value reduces computational burden & vice versa.
+//#define DEFAULT_SEQUENCE_MS 40
+#define DEFAULT_SEQUENCE_MS USE_AUTO_SEQUENCE_LEN
+
+/// Giving this value for the sequence length sets automatic parameter value
+/// according to tempo setting (recommended)
+#define USE_AUTO_SEQUENCE_LEN 0
+
+/// Seeking window default length in milliseconds for algorithm that finds the best possible
+/// overlapping location. This determines from how wide window the algorithm may look for an
+/// optimal joining location when mixing the sound sequences back together.
+///
+/// The bigger this window setting is, the higher the possibility to find a better mixing
+/// position will become, but at the same time large values may cause a "drifting" artifact
+/// because consequent sequences will be taken at more uneven intervals.
+///
+/// If there's a disturbing artifact that sounds as if a constant frequency was drifting
+/// around, try reducing this setting.
+///
+/// Increasing this value increases computational burden & vice versa.
+//#define DEFAULT_SEEKWINDOW_MS 15
+#define DEFAULT_SEEKWINDOW_MS USE_AUTO_SEEKWINDOW_LEN
+
+/// Giving this value for the seek window length sets automatic parameter value
+/// according to tempo setting (recommended)
+#define USE_AUTO_SEEKWINDOW_LEN 0
+
+/// Overlap length in milliseconds. When the chopped sound sequences are mixed back together,
+/// to form a continuous sound stream, this parameter defines over how long period the two
+/// consecutive sequences are let to overlap each other.
+///
+/// This shouldn't be that critical parameter. If you reduce the DEFAULT_SEQUENCE_MS setting
+/// by a large amount, you might wish to try a smaller value on this.
+///
+/// Increasing this value increases computational burden & vice versa.
+#define DEFAULT_OVERLAP_MS 8
+
+
+/// Class that does the time-stretch (tempo change) effect for the processed
+/// sound.
+class TDStretch : public FIFOProcessor
+{
+protected:
+ int channels;
+ int sampleReq;
+
+ int overlapLength;
+ int seekLength;
+ int seekWindowLength;
+ int overlapDividerBitsNorm;
+ int overlapDividerBitsPure;
+ int slopingDivider;
+ int sampleRate;
+ int sequenceMs;
+ int seekWindowMs;
+ int overlapMs;
+
+ unsigned long maxnorm;
+ float maxnormf;
+
+ double tempo;
+ double nominalSkip;
+ double skipFract;
+
+ bool bQuickSeek;
+ bool bAutoSeqSetting;
+ bool bAutoSeekSetting;
+ bool isBeginning;
+
+ SAMPLETYPE *pMidBuffer;
+ SAMPLETYPE *pMidBufferUnaligned;
+
+ FIFOSampleBuffer outputBuffer;
+ FIFOSampleBuffer inputBuffer;
+
+ void acceptNewOverlapLength(int newOverlapLength);
+
+ virtual void clearCrossCorrState();
+ void calculateOverlapLength(int overlapMs);
+
+ virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
+ virtual double calcCrossCorrAccumulate(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
+
+ virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
+ virtual int seekBestOverlapPositionQuick(const SAMPLETYPE *refPos);
+ virtual int seekBestOverlapPosition(const SAMPLETYPE *refPos);
+
+ virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+ virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+ virtual void overlapMulti(SAMPLETYPE *output, const SAMPLETYPE *input) const;
+
+ void clearMidBuffer();
+ void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
+
+ void calcSeqParameters();
+ void adaptNormalizer();
+
+ /// Changes the tempo of the given sound samples.
+ /// Returns amount of samples returned in the "output" buffer.
+ /// The maximum amount of samples that can be returned at a time is set by
+ /// the 'set_returnBuffer_size' function.
+ void processSamples();
+
+public:
+ TDStretch();
+ virtual ~TDStretch();
+
+ /// Operator 'new' is overloaded so that it automatically creates a suitable instance
+ /// depending on if we've a MMX/SSE/etc-capable CPU available or not.
+ static void *operator new(size_t s);
+
+ /// Use this function instead of "new" operator to create a new instance of this class.
+ /// This function automatically chooses a correct feature set depending on if the CPU
+ /// supports MMX/SSE/etc extensions.
+ static TDStretch *newInstance();
+
+ /// Returns the output buffer object
+ FIFOSamplePipe *getOutput() { return &outputBuffer; };
+
+ /// Returns the input buffer object
+ FIFOSamplePipe *getInput() { return &inputBuffer; };
+
+ /// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
+ /// tempo, larger faster tempo.
+ void setTempo(double newTempo);
+
+ /// Returns nonzero if there aren't any samples available for outputting.
+ virtual void clear();
+
+ /// Clears the input buffer
+ void clearInput();
+
+ /// Sets the number of channels, 1 = mono, 2 = stereo
+ void setChannels(int numChannels);
+
+ /// Enables/disables the quick position seeking algorithm. Zero to disable,
+ /// nonzero to enable
+ void enableQuickSeek(bool enable);
+
+ /// Returns nonzero if the quick seeking algorithm is enabled.
+ bool isQuickSeekEnabled() const;
+
+ /// Sets routine control parameters. These control are certain time constants
+ /// defining how the sound is stretched to the desired duration.
+ //
+ /// 'sampleRate' = sample rate of the sound
+ /// 'sequenceMS' = one processing sequence length in milliseconds
+ /// 'seekwindowMS' = seeking window length for scanning the best overlapping
+ /// position
+ /// 'overlapMS' = overlapping length
+ void setParameters(int sampleRate, ///< Samplerate of sound being processed (Hz)
+ int sequenceMS = -1, ///< Single processing sequence length (ms)
+ int seekwindowMS = -1, ///< Offset seeking window length (ms)
+ int overlapMS = -1 ///< Sequence overlapping length (ms)
+ );
+
+ /// Get routine control parameters, see setParameters() function.
+ /// Any of the parameters to this function can be NULL, in such case corresponding parameter
+ /// value isn't returned.
+ void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;
+
+ /// Adds 'numsamples' pcs of samples from the 'samples' memory position into
+ /// the input of the object.
+ virtual void putSamples(
+ const SAMPLETYPE *samples, ///< Input sample data
+ uint numSamples ///< Number of samples in 'samples' so that one sample
+ ///< contains both channels if stereo
+ );
+
+ /// return nominal input sample requirement for triggering a processing batch
+ int getInputSampleReq() const
+ {
+ return (int)(nominalSkip + 0.5);
+ }
+
+ /// return nominal output sample amount when running a processing batch
+ int getOutputBatchSize() const
+ {
+ return seekWindowLength - overlapLength;
+ }
+
+ /// return approximate initial input-output latency
+ int getLatency() const
+ {
+ return sampleReq;
+ }
+};
+
+
+// Implementation-specific class declarations:
+
+#ifdef SOUNDTOUCH_ALLOW_MMX
+ /// Class that implements MMX optimized routines for 16bit integer samples type.
+ class TDStretchMMX : public TDStretch
+ {
+ protected:
+ double calcCrossCorr(const short *mixingPos, const short *compare, double &norm);
+ double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm);
+ virtual void overlapStereo(short *output, const short *input) const;
+ virtual void clearCrossCorrState();
+ };
+#endif /// SOUNDTOUCH_ALLOW_MMX
+
+
+#ifdef SOUNDTOUCH_ALLOW_SSE
+ /// Class that implements SSE optimized routines for floating point samples type.
+ class TDStretchSSE : public TDStretch
+ {
+ protected:
+ double calcCrossCorr(const float *mixingPos, const float *compare, double &norm);
+ double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm);
+ };
+
+#endif /// SOUNDTOUCH_ALLOW_SSE
+
+}
+#endif /// TDStretch_H
diff --git a/media/libsoundtouch/src/cpu_detect.h b/media/libsoundtouch/src/cpu_detect.h
new file mode 100644
index 0000000000..093b6097aa
--- /dev/null
+++ b/media/libsoundtouch/src/cpu_detect.h
@@ -0,0 +1,55 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// A header file for detecting the Intel MMX instructions set extension.
+///
+/// Please see 'mmx_win.cpp', 'mmx_cpp.cpp' and 'mmx_non_x86.cpp' for the
+/// routine implementations for x86 Windows, x86 gnu version and non-x86
+/// platforms, respectively.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#ifndef _CPU_DETECT_H_
+#define _CPU_DETECT_H_
+
+#include "STTypes.h"
+
+#define SUPPORT_MMX 0x0001
+#define SUPPORT_3DNOW 0x0002
+#define SUPPORT_ALTIVEC 0x0004
+#define SUPPORT_SSE 0x0008
+#define SUPPORT_SSE2 0x0010
+
+/// Checks which instruction set extensions are supported by the CPU.
+///
+/// \return A bitmask of supported extensions, see SUPPORT_... defines.
+uint detectCPUextensions(void);
+
+/// Disables given set of instruction extensions. See SUPPORT_... defines.
+void disableExtensions(uint wDisableMask);
+
+#endif // _CPU_DETECT_H_
diff --git a/media/libsoundtouch/src/cpu_detect_x86.cpp b/media/libsoundtouch/src/cpu_detect_x86.cpp
new file mode 100644
index 0000000000..f0f026b20d
--- /dev/null
+++ b/media/libsoundtouch/src/cpu_detect_x86.cpp
@@ -0,0 +1,130 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// Generic version of the x86 CPU extension detection routine.
+///
+/// This file is for GNU & other non-Windows compilers, see 'cpu_detect_x86_win.cpp'
+/// for the Microsoft compiler version.
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "cpu_detect.h"
+#include "STTypes.h"
+
+
+#if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
+ #if defined(__GNUC__) && defined(HAVE_CPUID_H)
+ // gcc and clang
+ #include "cpuid.h"
+ #elif defined(_M_IX86)
+ // windows non-gcc
+ #include <intrin.h>
+ #endif
+
+ #define bit_MMX (1 << 23)
+ #define bit_SSE (1 << 25)
+ #define bit_SSE2 (1 << 26)
+#endif
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// processor instructions extension detection routines
+//
+//////////////////////////////////////////////////////////////////////////////
+
+// Flag variable indicating whick ISA extensions are disabled (for debugging)
+static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
+
+// Disables given set of instruction extensions. See SUPPORT_... defines.
+void disableExtensions(uint dwDisableMask)
+{
+ _dwDisabledISA = dwDisableMask;
+}
+
+
+/// Checks which instruction set extensions are supported by the CPU.
+uint detectCPUextensions(void)
+{
+/// If building for a 64bit system (no Itanium) and the user wants optimizations.
+/// Return the OR of SUPPORT_{MMX,SSE,SSE2}. 11001 or 0x19.
+/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
+#if ((defined(__GNUC__) && defined(__x86_64__)) \
+ || defined(_M_X64)) \
+ && defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
+ return 0x19 & ~_dwDisabledISA;
+
+/// If building for a 32bit system and the user wants optimizations.
+/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
+#elif ((defined(__GNUC__) && defined(__i386__)) \
+ || defined(_M_IX86)) \
+ && defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
+
+ if (_dwDisabledISA == 0xffffffff) return 0;
+
+ uint res = 0;
+
+#if !defined(__GNUC__)
+ // Window / VS version of cpuid. Notice that Visual Studio 2005 or later required
+ // for __cpuid intrinsic support.
+ int reg[4] = {-1};
+
+ // Check if no cpuid support.
+ __cpuid(reg,0);
+ if ((unsigned int)reg[0] == 0) return 0; // always disable extensions.
+
+ __cpuid(reg,1);
+ if ((unsigned int)reg[3] & bit_MMX) res = res | SUPPORT_MMX;
+ if ((unsigned int)reg[3] & bit_SSE) res = res | SUPPORT_SSE;
+ if ((unsigned int)reg[3] & bit_SSE2) res = res | SUPPORT_SSE2;
+#elif defined(HAVE_CPUID_H)
+ // GCC version of cpuid. Requires GCC 4.3.0 or later for __cpuid intrinsic support.
+ uint eax, ebx, ecx, edx; // unsigned int is the standard type. uint is defined by the compiler and not guaranteed to be portable.
+
+ // Check if no cpuid support.
+ if (!__get_cpuid (1, &eax, &ebx, &ecx, &edx)) return 0; // always disable extensions.
+
+ if (edx & bit_MMX) res = res | SUPPORT_MMX;
+ if (edx & bit_SSE) res = res | SUPPORT_SSE;
+ if (edx & bit_SSE2) res = res | SUPPORT_SSE2;
+#else
+ // Compatible with GCC but no cpuid.h.
+ return 0;
+#endif
+
+ return res & ~_dwDisabledISA;
+
+#else
+
+/// One of these is true:
+/// 1) We don't want optimizations.
+/// 2) Using an unsupported compiler.
+/// 3) Running on a non-x86 platform.
+ return 0;
+
+#endif
+}
diff --git a/media/libsoundtouch/src/mmx_optimized.cpp b/media/libsoundtouch/src/mmx_optimized.cpp
new file mode 100644
index 0000000000..0bc7fe86f7
--- /dev/null
+++ b/media/libsoundtouch/src/mmx_optimized.cpp
@@ -0,0 +1,396 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// MMX optimized routines. All MMX optimized functions have been gathered into
+/// this single source code file, regardless to their class or original source
+/// code file, in order to ease porting the library to other compiler and
+/// processor platforms.
+///
+/// The MMX-optimizations are programmed using MMX compiler intrinsics that
+/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
+/// should compile with both toolsets.
+///
+/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
+/// 6.0 processor pack" update to support compiler intrinsic syntax. The update
+/// is available for download at Microsoft Developers Network, see here:
+/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "STTypes.h"
+
+#ifdef SOUNDTOUCH_ALLOW_MMX
+// MMX routines available only with integer sample type
+
+using namespace soundtouch;
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of MMX optimized functions of class 'TDStretchMMX'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "TDStretch.h"
+#include <mmintrin.h>
+#include <limits.h>
+#include <math.h>
+
+
+// Calculates cross correlation of two buffers
+double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &dnorm)
+{
+ const __m64 *pVec1, *pVec2;
+ __m64 shifter;
+ __m64 accu, normaccu;
+ long corr, norm;
+ int i;
+
+ pVec1 = (__m64*)pV1;
+ pVec2 = (__m64*)pV2;
+
+ shifter = _m_from_int(overlapDividerBitsNorm);
+ normaccu = accu = _mm_setzero_si64();
+
+ // Process 4 parallel sets of 2 * stereo samples or 4 * mono samples
+ // during each round for improved CPU-level parallellization.
+ for (i = 0; i < channels * overlapLength / 16; i ++)
+ {
+ __m64 temp, temp2;
+
+ // dictionary of instructions:
+ // _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
+ // _mm_add_pi32 : 2*32bit add
+ // _m_psrad : 32bit right-shift
+
+ temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
+ _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
+ temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]), shifter),
+ _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec1[1]), shifter));
+ accu = _mm_add_pi32(accu, temp);
+ normaccu = _mm_add_pi32(normaccu, temp2);
+
+ temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
+ _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
+ temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]), shifter),
+ _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec1[3]), shifter));
+ accu = _mm_add_pi32(accu, temp);
+ normaccu = _mm_add_pi32(normaccu, temp2);
+
+ pVec1 += 4;
+ pVec2 += 4;
+ }
+
+ // copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
+ // and finally store the result into the variable "corr"
+
+ accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));
+ corr = _m_to_int(accu);
+
+ normaccu = _mm_add_pi32(normaccu, _mm_srli_si64(normaccu, 32));
+ norm = _m_to_int(normaccu);
+
+ // Clear MMS state
+ _m_empty();
+
+ if (norm > (long)maxnorm)
+ {
+ // modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode
+ #pragma omp critical
+ if (norm > (long)maxnorm)
+ {
+ maxnorm = norm;
+ }
+ }
+
+ // Normalize result by dividing by sqrt(norm) - this step is easiest
+ // done using floating point operation
+ dnorm = (double)norm;
+
+ return (double)corr / sqrt(dnorm < 1e-9 ? 1.0 : dnorm);
+ // Note: Warning about the missing EMMS instruction is harmless
+ // as it'll be called elsewhere.
+}
+
+
+/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
+double TDStretchMMX::calcCrossCorrAccumulate(const short *pV1, const short *pV2, double &dnorm)
+{
+ const __m64 *pVec1, *pVec2;
+ __m64 shifter;
+ __m64 accu;
+ long corr, lnorm;
+ int i;
+
+ // cancel first normalizer tap from previous round
+ lnorm = 0;
+ for (i = 1; i <= channels; i ++)
+ {
+ lnorm -= (pV1[-i] * pV1[-i]) >> overlapDividerBitsNorm;
+ }
+
+ pVec1 = (__m64*)pV1;
+ pVec2 = (__m64*)pV2;
+
+ shifter = _m_from_int(overlapDividerBitsNorm);
+ accu = _mm_setzero_si64();
+
+ // Process 4 parallel sets of 2 * stereo samples or 4 * mono samples
+ // during each round for improved CPU-level parallellization.
+ for (i = 0; i < channels * overlapLength / 16; i ++)
+ {
+ __m64 temp;
+
+ // dictionary of instructions:
+ // _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
+ // _mm_add_pi32 : 2*32bit add
+ // _m_psrad : 32bit right-shift
+
+ temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
+ _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
+ accu = _mm_add_pi32(accu, temp);
+
+ temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
+ _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
+ accu = _mm_add_pi32(accu, temp);
+
+ pVec1 += 4;
+ pVec2 += 4;
+ }
+
+ // copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
+ // and finally store the result into the variable "corr"
+
+ accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));
+ corr = _m_to_int(accu);
+
+ // Clear MMS state
+ _m_empty();
+
+ // update normalizer with last samples of this round
+ pV1 = (short *)pVec1;
+ for (int j = 1; j <= channels; j ++)
+ {
+ lnorm += (pV1[-j] * pV1[-j]) >> overlapDividerBitsNorm;
+ }
+ dnorm += (double)lnorm;
+
+ if (lnorm > (long)maxnorm)
+ {
+ maxnorm = lnorm;
+ }
+
+ // Normalize result by dividing by sqrt(norm) - this step is easiest
+ // done using floating point operation
+ return (double)corr / sqrt((dnorm < 1e-9) ? 1.0 : dnorm);
+}
+
+
+void TDStretchMMX::clearCrossCorrState()
+{
+ // Clear MMS state
+ _m_empty();
+ //_asm EMMS;
+}
+
+
+// MMX-optimized version of the function overlapStereo
+void TDStretchMMX::overlapStereo(short *output, const short *input) const
+{
+ const __m64 *pVinput, *pVMidBuf;
+ __m64 *pVdest;
+ __m64 mix1, mix2, adder, shifter;
+ int i;
+
+ pVinput = (const __m64*)input;
+ pVMidBuf = (const __m64*)pMidBuffer;
+ pVdest = (__m64*)output;
+
+ // mix1 = mixer values for 1st stereo sample
+ // mix1 = mixer values for 2nd stereo sample
+ // adder = adder for updating mixer values after each round
+
+ mix1 = _mm_set_pi16(0, overlapLength, 0, overlapLength);
+ adder = _mm_set_pi16(1, -1, 1, -1);
+ mix2 = _mm_add_pi16(mix1, adder);
+ adder = _mm_add_pi16(adder, adder);
+
+ // Overlaplength-division by shifter. "+1" is to account for "-1" deduced in
+ // overlapDividerBits calculation earlier.
+ shifter = _m_from_int(overlapDividerBitsPure + 1);
+
+ for (i = 0; i < overlapLength / 4; i ++)
+ {
+ __m64 temp1, temp2;
+
+ // load & shuffle data so that input & mixbuffer data samples are paired
+ temp1 = _mm_unpacklo_pi16(pVMidBuf[0], pVinput[0]); // = i0l m0l i0r m0r
+ temp2 = _mm_unpackhi_pi16(pVMidBuf[0], pVinput[0]); // = i1l m1l i1r m1r
+
+ // temp = (temp .* mix) >> shifter
+ temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
+ temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
+ pVdest[0] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
+
+ // update mix += adder
+ mix1 = _mm_add_pi16(mix1, adder);
+ mix2 = _mm_add_pi16(mix2, adder);
+
+ // --- second round begins here ---
+
+ // load & shuffle data so that input & mixbuffer data samples are paired
+ temp1 = _mm_unpacklo_pi16(pVMidBuf[1], pVinput[1]); // = i2l m2l i2r m2r
+ temp2 = _mm_unpackhi_pi16(pVMidBuf[1], pVinput[1]); // = i3l m3l i3r m3r
+
+ // temp = (temp .* mix) >> shifter
+ temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
+ temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
+ pVdest[1] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
+
+ // update mix += adder
+ mix1 = _mm_add_pi16(mix1, adder);
+ mix2 = _mm_add_pi16(mix2, adder);
+
+ pVinput += 2;
+ pVMidBuf += 2;
+ pVdest += 2;
+ }
+
+ _m_empty(); // clear MMS state
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of MMX optimized functions of class 'FIRFilter'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "FIRFilter.h"
+
+
+FIRFilterMMX::FIRFilterMMX() : FIRFilter()
+{
+ filterCoeffsAlign = NULL;
+ filterCoeffsUnalign = NULL;
+}
+
+
+FIRFilterMMX::~FIRFilterMMX()
+{
+ delete[] filterCoeffsUnalign;
+}
+
+
+// (overloaded) Calculates filter coefficients for MMX routine
+void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor)
+{
+ uint i;
+ FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
+
+ // Ensure that filter coeffs array is aligned to 16-byte boundary
+ delete[] filterCoeffsUnalign;
+ filterCoeffsUnalign = new short[2 * newLength + 8];
+ filterCoeffsAlign = (short *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
+
+ // rearrange the filter coefficients for mmx routines
+ for (i = 0;i < length; i += 4)
+ {
+ filterCoeffsAlign[2 * i + 0] = coeffs[i + 0];
+ filterCoeffsAlign[2 * i + 1] = coeffs[i + 2];
+ filterCoeffsAlign[2 * i + 2] = coeffs[i + 0];
+ filterCoeffsAlign[2 * i + 3] = coeffs[i + 2];
+
+ filterCoeffsAlign[2 * i + 4] = coeffs[i + 1];
+ filterCoeffsAlign[2 * i + 5] = coeffs[i + 3];
+ filterCoeffsAlign[2 * i + 6] = coeffs[i + 1];
+ filterCoeffsAlign[2 * i + 7] = coeffs[i + 3];
+ }
+}
+
+
+// mmx-optimized version of the filter routine for stereo sound
+uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
+{
+ // Create stack copies of the needed member variables for asm routines :
+ uint i, j;
+ __m64 *pVdest = (__m64*)dest;
+
+ if (length < 2) return 0;
+
+ for (i = 0; i < (numSamples - length) / 2; i ++)
+ {
+ __m64 accu1;
+ __m64 accu2;
+ const __m64 *pVsrc = (const __m64*)src;
+ const __m64 *pVfilter = (const __m64*)filterCoeffsAlign;
+
+ accu1 = accu2 = _mm_setzero_si64();
+ for (j = 0; j < lengthDiv8 * 2; j ++)
+ {
+ __m64 temp1, temp2;
+
+ temp1 = _mm_unpacklo_pi16(pVsrc[0], pVsrc[1]); // = l2 l0 r2 r0
+ temp2 = _mm_unpackhi_pi16(pVsrc[0], pVsrc[1]); // = l3 l1 r3 r1
+
+ accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp1, pVfilter[0])); // += l2*f2+l0*f0 r2*f2+r0*f0
+ accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp2, pVfilter[1])); // += l3*f3+l1*f1 r3*f3+r1*f1
+
+ temp1 = _mm_unpacklo_pi16(pVsrc[1], pVsrc[2]); // = l4 l2 r4 r2
+
+ accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp2, pVfilter[0])); // += l3*f2+l1*f0 r3*f2+r1*f0
+ accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp1, pVfilter[1])); // += l4*f3+l2*f1 r4*f3+r2*f1
+
+ // accu1 += l2*f2+l0*f0 r2*f2+r0*f0
+ // += l3*f3+l1*f1 r3*f3+r1*f1
+
+ // accu2 += l3*f2+l1*f0 r3*f2+r1*f0
+ // l4*f3+l2*f1 r4*f3+r2*f1
+
+ pVfilter += 2;
+ pVsrc += 2;
+ }
+ // accu >>= resultDivFactor
+ accu1 = _mm_srai_pi32(accu1, resultDivFactor);
+ accu2 = _mm_srai_pi32(accu2, resultDivFactor);
+
+ // pack 2*2*32bits => 4*16 bits
+ pVdest[0] = _mm_packs_pi32(accu1, accu2);
+ src += 4;
+ pVdest ++;
+ }
+
+ _m_empty(); // clear emms state
+
+ return (numSamples & 0xfffffffe) - length;
+}
+
+#else
+
+// workaround to not complain about empty module
+bool _dontcomplain_mmx_empty;
+
+#endif // SOUNDTOUCH_ALLOW_MMX
diff --git a/media/libsoundtouch/src/moz.build b/media/libsoundtouch/src/moz.build
new file mode 100644
index 0000000000..63173892f2
--- /dev/null
+++ b/media/libsoundtouch/src/moz.build
@@ -0,0 +1,53 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+EXPORTS.soundtouch += [
+ 'FIFOSamplePipe.h',
+ 'SoundTouch.h',
+ 'soundtouch_config.h',
+ 'SoundTouchFactory.h',
+ 'STTypes.h',
+]
+
+UNIFIED_SOURCES += [
+ 'AAFilter.cpp',
+ 'cpu_detect_x86.cpp',
+ 'FIFOSampleBuffer.cpp',
+ 'FIRFilter.cpp',
+ 'InterpolateCubic.cpp',
+ 'InterpolateLinear.cpp',
+ 'InterpolateShannon.cpp',
+ 'RateTransposer.cpp',
+ 'SoundTouch.cpp',
+ 'SoundTouchFactory.cpp',
+ 'TDStretch.cpp',
+]
+
+if CONFIG['INTEL_ARCHITECTURE']:
+ if CONFIG['MOZ_SAMPLE_TYPE_FLOAT32']:
+ SOURCES += ['sse_optimized.cpp']
+ SOURCES['sse_optimized.cpp'].flags += CONFIG['SSE2_FLAGS']
+ else:
+ SOURCES += ['mmx_optimized.cpp']
+ SOURCES['mmx_optimized.cpp'].flags += CONFIG['MMX_FLAGS']
+
+if CONFIG['OS_ARCH'] != 'WINNT':
+ # GCC/Clang require permissions to be explicitly set for the soundtouch
+ # header.
+ CXXFLAGS += ['-include', SRCDIR + '/soundtouch_perms.h']
+else:
+ # Windows need alloca renamed to _alloca
+ DEFINES['alloca'] = '_alloca'
+
+DEFINES['BUILDING_SOUNDTOUCH'] = 1
+
+# We allow warnings for third-party code that can be updated from upstream.
+AllowCompilerWarnings()
+
+FINAL_LIBRARY = 'lgpllibs'
+
+# Use abort() instead of exception in SoundTouch.
+DEFINES['ST_NO_EXCEPTION_HANDLING'] = 1
diff --git a/media/libsoundtouch/src/soundtouch_config.h b/media/libsoundtouch/src/soundtouch_config.h
new file mode 100644
index 0000000000..469bb822af
--- /dev/null
+++ b/media/libsoundtouch/src/soundtouch_config.h
@@ -0,0 +1,7 @@
+#include "mozilla/SSE.h"
+
+#ifdef MOZ_SAMPLE_TYPE_FLOAT32
+#define SOUNDTOUCH_FLOAT_SAMPLES 1
+#else
+#define SOUNDTOUCH_INTEGER_SAMPLES 1
+#endif
diff --git a/media/libsoundtouch/src/soundtouch_perms.h b/media/libsoundtouch/src/soundtouch_perms.h
new file mode 100644
index 0000000000..0af2fe6183
--- /dev/null
+++ b/media/libsoundtouch/src/soundtouch_perms.h
@@ -0,0 +1,18 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Include file for fixing symbol visibility on non-windows platforms, until
+// system headers wrappers work uniformly across all of them.
+
+#ifndef MOZILLA_SOUNDTOUCH_PERMS_H
+#define MOZILLA_SOUNDTOUCH_PERMS_H
+
+#pragma GCC visibility push(default)
+#include "SoundTouch.h"
+#include "SoundTouchFactory.h"
+#pragma GCC visibility pop
+
+#endif // MOZILLA_SOUNDTOUCH_PERMS_H
diff --git a/media/libsoundtouch/src/sse_optimized.cpp b/media/libsoundtouch/src/sse_optimized.cpp
new file mode 100644
index 0000000000..9c16ea8f89
--- /dev/null
+++ b/media/libsoundtouch/src/sse_optimized.cpp
@@ -0,0 +1,365 @@
+////////////////////////////////////////////////////////////////////////////////
+///
+/// SSE optimized routines for Pentium-III, Athlon-XP and later CPUs. All SSE
+/// optimized functions have been gathered into this single source
+/// code file, regardless to their class or original source code file, in order
+/// to ease porting the library to other compiler and processor platforms.
+///
+/// The SSE-optimizations are programmed using SSE compiler intrinsics that
+/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
+/// should compile with both toolsets.
+///
+/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
+/// 6.0 processor pack" update to support SSE instruction set. The update is
+/// available for download at Microsoft Developers Network, see here:
+/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
+///
+/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
+/// perform a search with keywords "processor pack".
+///
+/// Author : Copyright (c) Olli Parviainen
+/// Author e-mail : oparviai 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// License :
+//
+// SoundTouch audio processing library
+// Copyright (c) Olli Parviainen
+//
+// This library is free software; you can redistribute it and/or
+// modify it under the terms of the GNU Lesser General Public
+// License as published by the Free Software Foundation; either
+// version 2.1 of the License, or (at your option) any later version.
+//
+// This library is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+// Lesser General Public License for more details.
+//
+// You should have received a copy of the GNU Lesser General Public
+// License along with this library; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+////////////////////////////////////////////////////////////////////////////////
+
+#include "cpu_detect.h"
+#include "STTypes.h"
+
+using namespace soundtouch;
+
+#ifdef SOUNDTOUCH_ALLOW_SSE
+
+// SSE routines available only with float sample type
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of SSE optimized functions of class 'TDStretchSSE'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "TDStretch.h"
+#include <xmmintrin.h>
+#include <math.h>
+
+// Calculates cross correlation of two buffers
+double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &anorm)
+{
+ int i;
+ const float *pVec1;
+ const __m128 *pVec2;
+ __m128 vSum, vNorm;
+
+ // Note. It means a major slow-down if the routine needs to tolerate
+ // unaligned __m128 memory accesses. It's way faster if we can skip
+ // unaligned slots and use _mm_load_ps instruction instead of _mm_loadu_ps.
+ // This can mean up to ~ 10-fold difference (incl. part of which is
+ // due to skipping every second round for stereo sound though).
+ //
+ // Compile-time define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
+ // for choosing if this little cheating is allowed.
+
+#ifdef ST_SIMD_AVOID_UNALIGNED
+ // Little cheating allowed, return valid correlation only for
+ // aligned locations, meaning every second round for stereo sound.
+
+ #define _MM_LOAD _mm_load_ps
+
+ if (((ulongptr)pV1) & 15) return -1e50; // skip unaligned locations
+
+#else
+ // No cheating allowed, use unaligned load & take the resulting
+ // performance hit.
+ #define _MM_LOAD _mm_loadu_ps
+#endif
+
+ // ensure overlapLength is divisible by 8
+ assert((overlapLength % 8) == 0);
+
+ // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
+ // Note: pV2 _must_ be aligned to 16-bit boundary, pV1 need not.
+ pVec1 = (const float*)pV1;
+ pVec2 = (const __m128*)pV2;
+ vSum = vNorm = _mm_setzero_ps();
+
+ // Unroll the loop by factor of 4 * 4 operations. Use same routine for
+ // stereo & mono, for mono it just means twice the amount of unrolling.
+ for (i = 0; i < channels * overlapLength / 16; i ++)
+ {
+ __m128 vTemp;
+ // vSum += pV1[0..3] * pV2[0..3]
+ vTemp = _MM_LOAD(pVec1);
+ vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp ,pVec2[0]));
+ vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+ // vSum += pV1[4..7] * pV2[4..7]
+ vTemp = _MM_LOAD(pVec1 + 4);
+ vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[1]));
+ vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+ // vSum += pV1[8..11] * pV2[8..11]
+ vTemp = _MM_LOAD(pVec1 + 8);
+ vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[2]));
+ vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+ // vSum += pV1[12..15] * pV2[12..15]
+ vTemp = _MM_LOAD(pVec1 + 12);
+ vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[3]));
+ vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
+
+ pVec1 += 16;
+ pVec2 += 4;
+ }
+
+ // return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]
+ float *pvNorm = (float*)&vNorm;
+ float norm = (pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
+ anorm = norm;
+
+ float *pvSum = (float*)&vSum;
+ return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / sqrt(norm < 1e-9 ? 1.0 : norm);
+
+ /* This is approximately corresponding routine in C-language yet without normalization:
+ double corr, norm;
+ uint i;
+
+ // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
+ corr = norm = 0.0;
+ for (i = 0; i < channels * overlapLength / 16; i ++)
+ {
+ corr += pV1[0] * pV2[0] +
+ pV1[1] * pV2[1] +
+ pV1[2] * pV2[2] +
+ pV1[3] * pV2[3] +
+ pV1[4] * pV2[4] +
+ pV1[5] * pV2[5] +
+ pV1[6] * pV2[6] +
+ pV1[7] * pV2[7] +
+ pV1[8] * pV2[8] +
+ pV1[9] * pV2[9] +
+ pV1[10] * pV2[10] +
+ pV1[11] * pV2[11] +
+ pV1[12] * pV2[12] +
+ pV1[13] * pV2[13] +
+ pV1[14] * pV2[14] +
+ pV1[15] * pV2[15];
+
+ for (j = 0; j < 15; j ++) norm += pV1[j] * pV1[j];
+
+ pV1 += 16;
+ pV2 += 16;
+ }
+ return corr / sqrt(norm);
+ */
+}
+
+
+
+double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2, double &norm)
+{
+ // call usual calcCrossCorr function because SSE does not show big benefit of
+ // accumulating "norm" value, and also the "norm" rolling algorithm would get
+ // complicated due to SSE-specific alignment-vs-nonexact correlation rules.
+ return calcCrossCorr(pV1, pV2, norm);
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// implementation of SSE optimized functions of class 'FIRFilter'
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#include "FIRFilter.h"
+
+FIRFilterSSE::FIRFilterSSE() : FIRFilter()
+{
+ filterCoeffsAlign = NULL;
+ filterCoeffsUnalign = NULL;
+}
+
+
+FIRFilterSSE::~FIRFilterSSE()
+{
+ delete[] filterCoeffsUnalign;
+ filterCoeffsAlign = NULL;
+ filterCoeffsUnalign = NULL;
+}
+
+
+// (overloaded) Calculates filter coefficients for SSE routine
+void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
+{
+ uint i;
+ float fDivider;
+
+ FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
+
+ // Scale the filter coefficients so that it won't be necessary to scale the filtering result
+ // also rearrange coefficients suitably for SSE
+ // Ensure that filter coeffs array is aligned to 16-byte boundary
+ delete[] filterCoeffsUnalign;
+ filterCoeffsUnalign = new float[2 * newLength + 4];
+ filterCoeffsAlign = (float *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
+
+ fDivider = (float)resultDivider;
+
+ // rearrange the filter coefficients for mmx routines
+ for (i = 0; i < newLength; i ++)
+ {
+ filterCoeffsAlign[2 * i + 0] =
+ filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
+ }
+}
+
+
+
+// SSE-optimized version of the filter routine for stereo sound
+uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint numSamples) const
+{
+ int count = (int)((numSamples - length) & (uint)-2);
+ int j;
+
+ assert(count % 2 == 0);
+
+ if (count < 2) return 0;
+
+ assert(source != NULL);
+ assert(dest != NULL);
+ assert((length % 8) == 0);
+ assert(filterCoeffsAlign != NULL);
+ assert(((ulongptr)filterCoeffsAlign) % 16 == 0);
+
+ // filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
+ #pragma omp parallel for
+ for (j = 0; j < count; j += 2)
+ {
+ const float *pSrc;
+ float *pDest;
+ const __m128 *pFil;
+ __m128 sum1, sum2;
+ uint i;
+
+ pSrc = (const float*)source + j * 2; // source audio data
+ pDest = dest + j * 2; // destination audio data
+ pFil = (const __m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
+ // are aligned to 16-byte boundary
+ sum1 = sum2 = _mm_setzero_ps();
+
+ for (i = 0; i < length / 8; i ++)
+ {
+ // Unroll loop for efficiency & calculate filter for 2*2 stereo samples
+ // at each pass
+
+ // sum1 is accu for 2*2 filtered stereo sound data at the primary sound data offset
+ // sum2 is accu for 2*2 filtered stereo sound data for the next sound sample offset.
+
+ sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc) , pFil[0]));
+ sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 2), pFil[0]));
+
+ sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 4), pFil[1]));
+ sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 6), pFil[1]));
+
+ sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 8) , pFil[2]));
+ sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 10), pFil[2]));
+
+ sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 12), pFil[3]));
+ sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 14), pFil[3]));
+
+ pSrc += 16;
+ pFil += 4;
+ }
+
+ // Now sum1 and sum2 both have a filtered 2-channel sample each, but we still need
+ // to sum the two hi- and lo-floats of these registers together.
+
+ // post-shuffle & add the filtered values and store to dest.
+ _mm_storeu_ps(pDest, _mm_add_ps(
+ _mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)), // s2_1 s2_0 s1_3 s1_2
+ _mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0)) // s2_3 s2_2 s1_1 s1_0
+ ));
+ }
+
+ // Ideas for further improvement:
+ // 1. If it could be guaranteed that 'source' were always aligned to 16-byte
+ // boundary, a faster aligned '_mm_load_ps' instruction could be used.
+ // 2. If it could be guaranteed that 'dest' were always aligned to 16-byte
+ // boundary, a faster '_mm_store_ps' instruction could be used.
+
+ return (uint)count;
+
+ /* original routine in C-language. please notice the C-version has differently
+ organized coefficients though.
+ double suml1, suml2;
+ double sumr1, sumr2;
+ uint i, j;
+
+ for (j = 0; j < count; j += 2)
+ {
+ const float *ptr;
+ const float *pFil;
+
+ suml1 = sumr1 = 0.0;
+ suml2 = sumr2 = 0.0;
+ ptr = src;
+ pFil = filterCoeffs;
+ for (i = 0; i < lengthLocal; i ++)
+ {
+ // unroll loop for efficiency.
+
+ suml1 += ptr[0] * pFil[0] +
+ ptr[2] * pFil[2] +
+ ptr[4] * pFil[4] +
+ ptr[6] * pFil[6];
+
+ sumr1 += ptr[1] * pFil[1] +
+ ptr[3] * pFil[3] +
+ ptr[5] * pFil[5] +
+ ptr[7] * pFil[7];
+
+ suml2 += ptr[8] * pFil[0] +
+ ptr[10] * pFil[2] +
+ ptr[12] * pFil[4] +
+ ptr[14] * pFil[6];
+
+ sumr2 += ptr[9] * pFil[1] +
+ ptr[11] * pFil[3] +
+ ptr[13] * pFil[5] +
+ ptr[15] * pFil[7];
+
+ ptr += 16;
+ pFil += 8;
+ }
+ dest[0] = (float)suml1;
+ dest[1] = (float)sumr1;
+ dest[2] = (float)suml2;
+ dest[3] = (float)sumr2;
+
+ src += 4;
+ dest += 4;
+ }
+ */
+}
+
+#endif // SOUNDTOUCH_ALLOW_SSE
diff --git a/media/libsoundtouch/update.sh b/media/libsoundtouch/update.sh
new file mode 100755
index 0000000000..3894ee69f0
--- /dev/null
+++ b/media/libsoundtouch/update.sh
@@ -0,0 +1,73 @@
+#!/bin/bash
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+# Usage: ./update.sh <SoundTouch_src_directory>
+#
+# Copies the needed files from a directory containing the original
+# soundtouch sources that we need for HTML5 media playback rate change.
+
+cp $1/COPYING.TXT LICENSE
+cp $1/source/SoundTouch/AAFilter.cpp src
+cp $1/source/SoundTouch/AAFilter.h src
+cp $1/source/SoundTouch/cpu_detect.h src
+cp $1/source/SoundTouch/cpu_detect_x86.cpp src
+cp $1/source/SoundTouch/FIFOSampleBuffer.cpp src
+cp $1/source/SoundTouch/FIRFilter.cpp src
+cp $1/source/SoundTouch/FIRFilter.h src
+cp $1/source/SoundTouch/InterpolateLinear.cpp src
+cp $1/source/SoundTouch/InterpolateLinear.h src
+cp $1/source/SoundTouch/InterpolateCubic.cpp src
+cp $1/source/SoundTouch/InterpolateCubic.h src
+cp $1/source/SoundTouch/InterpolateShannon.cpp src
+cp $1/source/SoundTouch/InterpolateShannon.h src
+cp $1/source/SoundTouch/mmx_optimized.cpp src
+cp $1/source/SoundTouch/RateTransposer.cpp src
+cp $1/source/SoundTouch/RateTransposer.h src
+cp $1/source/SoundTouch/SoundTouch.cpp src
+cp $1/source/SoundTouch/sse_optimized.cpp src
+cp $1/source/SoundTouch/TDStretch.cpp src
+cp $1/source/SoundTouch/TDStretch.h src
+cp $1/include/SoundTouch.h src
+cp $1/include/FIFOSampleBuffer.h src
+cp $1/include/FIFOSamplePipe.h src
+cp $1/include/SoundTouch.h src
+cp $1/include/STTypes.h src
+
+# Remove the Windows line ending characters from the files.
+for i in src/*
+do
+ cat $i | tr -d '\015' > $i.lf
+ mv $i.lf $i
+done
+
+# Patch the imported files.
+patch -p1 < moz-libsoundtouch.patch
+
+if [ -d $1/.git ]; then
+ rev=$(cd $1 && git rev-parse --verify HEAD)
+ date=$(cd $1 && git show -s --format=%ci HEAD)
+ dirty=$(cd $1 && git diff-index --name-only HEAD)
+ set +e
+ pre_rev=$(grep -o '[[:xdigit:]]\{40\}' moz.yaml)
+ commits=$(cd $1 && git log --pretty=format:'%h - %s' $pre_rev..$rev)
+ set -e
+fi
+
+if [ -n "$rev" ]; then
+ version=$rev
+ if [ -n "$dirty" ]; then
+ version=$version-dirty
+ echo "WARNING: updating from a dirty git repository."
+ fi
+ sed -i.bak -e "s/^ *release:.*/ release: \"$version ($date)\"/" moz.yaml
+ if [[ ! "$( grep "$version" moz.yaml )" ]]; then
+ echo "Updating moz.yaml failed."
+ exit 1
+ fi
+ rm moz.yaml.bak
+ [[ -n "$commits" ]] && echo -e "Pick commits:\n$commits"
+else
+ echo "Remember to update moz.yaml with the version details."
+fi