diff --git a/b2g/installer/package-manifest.in b/b2g/installer/package-manifest.in index 8687a1eed2b0..9399b762c81f 100644 --- a/b2g/installer/package-manifest.in +++ b/b2g/installer/package-manifest.in @@ -54,7 +54,6 @@ @BINPATH@/@DLL_PREFIX@xpcom@DLL_SUFFIX@ @BINPATH@/@DLL_PREFIX@nspr4@DLL_SUFFIX@ @BINPATH@/@DLL_PREFIX@mozalloc@DLL_SUFFIX@ -@BINPATH@/@DLL_PREFIX@soundtouch@DLL_SUFFIX@ #ifdef XP_MACOSX @BINPATH@/XUL #else diff --git a/browser/installer/package-manifest.in b/browser/installer/package-manifest.in index 266b49dba0cd..52f3f4be6738 100644 --- a/browser/installer/package-manifest.in +++ b/browser/installer/package-manifest.in @@ -54,7 +54,6 @@ @BINPATH@/@DLL_PREFIX@gkmedias@DLL_SUFFIX@ #endif @BINPATH@/@DLL_PREFIX@mozalloc@DLL_SUFFIX@ -@BINPATH@/@DLL_PREFIX@soundtouch@DLL_SUFFIX@ #ifdef MOZ_SHARED_MOZGLUE @BINPATH@/@DLL_PREFIX@mozglue@DLL_SUFFIX@ #endif diff --git a/config/system-headers b/config/system-headers index 74fc247fdb05..f781c8220d08 100644 --- a/config/system-headers +++ b/config/system-headers @@ -981,8 +981,6 @@ plstr.h plarenas.h plarena.h plhash.h -speex/speex_resampler.h -soundtouch/SoundTouch.h #if MOZ_NATIVE_PNG==1 png.h #endif @@ -1055,6 +1053,7 @@ vpx/vpx_encoder.h vpx/vp8cx.h vpx/vp8dx.h sydneyaudio/sydney_audio.h +speex/speex_resampler.h vorbis/codec.h theora/theoradec.h tremor/ivorbiscodec.h diff --git a/configure.in b/configure.in index 5bb372e5a020..ef2acbc7a856 100644 --- a/configure.in +++ b/configure.in @@ -4225,7 +4225,6 @@ MOZ_OGG=1 MOZ_RAW= MOZ_SYDNEYAUDIO= MOZ_SPEEX_RESAMPLER=1 -MOZ_SOUNDTOUCH=1 MOZ_CUBEB= MOZ_VORBIS= MOZ_TREMOR= @@ -5581,19 +5580,6 @@ if test -n "$MOZ_SPEEX_RESAMPLER"; then AC_DEFINE(MOZ_SPEEX_RESAMPLER) fi -if test -n "$MOZ_SOUNDTOUCH"; then - AC_DEFINE(MOZ_SOUNDTOUCH) -fi - -if test -z "$GNU_CC" -a "$OS_ARCH" = "WINNT"; then - SOUNDTOUCH_LIBS='$(LIBXUL_DIST)/lib/$(LIB_PREFIX)soundtouch.$(LIB_SUFFIX)' -else - SOUNDTOUCH_LIBS='-lsoundtouch' -fi -AC_SUBST(SOUNDTOUCH_CFLAGS) -AC_SUBST(SOUNDTOUCH_LIBS) -AC_SUBST(SOUNDTOUCH_CONFIG) - if test -n "$MOZ_CUBEB"; then case "$target" in *-android*|*-linuxandroid*) @@ -8667,7 +8653,6 @@ AC_SUBST(MOZ_APP_EXTRA_LIBS) AC_SUBST(MOZ_MEDIA) AC_SUBST(MOZ_SYDNEYAUDIO) AC_SUBST(MOZ_SPEEX_RESAMPLER) -AC_SUBST(MOZ_SOUNDTOUCH) AC_SUBST(MOZ_CUBEB) AC_SUBST(MOZ_WAVE) AC_SUBST(MOZ_VORBIS) diff --git a/js/src/config/system-headers b/js/src/config/system-headers index 74fc247fdb05..f781c8220d08 100644 --- a/js/src/config/system-headers +++ b/js/src/config/system-headers @@ -981,8 +981,6 @@ plstr.h plarenas.h plarena.h plhash.h -speex/speex_resampler.h -soundtouch/SoundTouch.h #if MOZ_NATIVE_PNG==1 png.h #endif @@ -1055,6 +1053,7 @@ vpx/vpx_encoder.h vpx/vp8cx.h vpx/vp8dx.h sydneyaudio/sydney_audio.h +speex/speex_resampler.h vorbis/codec.h theora/theoradec.h tremor/ivorbiscodec.h diff --git a/media/libsoundtouch/AUTHORS b/media/libsoundtouch/AUTHORS deleted file mode 100644 index 666081e1952c..000000000000 --- a/media/libsoundtouch/AUTHORS +++ /dev/null @@ -1,4 +0,0 @@ -The SoundTouch Library -Copyright © Olli Parviainen 2001-2012 - -http://www.surina.net/soundtouch/ diff --git a/media/libsoundtouch/LICENSE b/media/libsoundtouch/LICENSE deleted file mode 100644 index 5b2161be204c..000000000000 --- a/media/libsoundtouch/LICENSE +++ /dev/null @@ -1,458 +0,0 @@ - GNU LESSER GENERAL PUBLIC LICENSE - Version 2.1, February 1999 - - Copyright (C) 1991, 1999 Free Software Foundation, Inc. - 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - Everyone is permitted to copy and distribute verbatim copies - of this license document, but changing it is not allowed. - -[This is the first released version of the Lesser GPL. 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If a copy of the MPL was not distributed with this -# file, You can obtain one at http://mozilla.org/MPL/2.0/. - -DEPTH = ../.. -topsrcdir = @top_srcdir@ -srcdir = @srcdir@ -VPATH = @srcdir@ - -include $(DEPTH)/config/autoconf.mk - -MODULE = soundtouch - -DIRS = src \ - $(NULL) - -include $(topsrcdir)/config/rules.mk diff --git a/media/libsoundtouch/README_MOZILLA b/media/libsoundtouch/README_MOZILLA deleted file mode 100644 index 141860babd42..000000000000 --- a/media/libsoundtouch/README_MOZILLA +++ /dev/null @@ -1,8 +0,0 @@ -These files are from the SoundTouch library (http://www.surina.net/soundtouch/), -and are extracted from the revision r143 of the svn repository at -https://soundtouch.svn.sourceforge.net/svnroot/soundtouch/trunk. - -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. - diff --git a/media/libsoundtouch/moz-libsoundtouch.patch b/media/libsoundtouch/moz-libsoundtouch.patch deleted file mode 100644 index f78ec818561d..000000000000 --- a/media/libsoundtouch/moz-libsoundtouch.patch +++ /dev/null @@ -1,509 +0,0 @@ -unchanged: ---- /src/STTypes.h 2012-08-02 10:04:06.301691592 -0700 -+++ /src/STTypes.h -@@ -42,21 +42,13 @@ - typedef unsigned int uint; - typedef unsigned long ulong; - --#ifdef __GNUC__ -- // In GCC, include soundtouch_config.h made by config scritps -- #include "soundtouch_config.h" --#endif -- --#ifndef _WINDEF_ -- // if these aren't defined already by Windows headers, define now -- -- typedef int BOOL; -- -- #define FALSE 0 -- #define TRUE 1 -- --#endif // _WINDEF_ -+#include "soundtouch_config.h" - -+#ifdef WIN32 -+#define EXPORT __declspec(dllexport) -+#else -+#define EXPORT -+#endif - - namespace soundtouch - { -@@ -82,7 +74,7 @@ - /// 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 -+ #define SOUNDTOUCH_FLOAT_SAMPLES 1 //< 32bit float samples - - #endif - -@@ -144,10 +136,10 @@ - - #endif // SOUNDTOUCH_INTEGER_SAMPLES - --}; -+} - - // define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions: --// #define ST_NO_EXCEPTION_HANDLING 1 -+#define ST_NO_EXCEPTION_HANDLING 1 - #ifdef ST_NO_EXCEPTION_HANDLING - // Exceptions disabled. Throw asserts instead if enabled. - #include ---- /src/SoundTouch.h 2012-08-02 10:04:06.301691592 -0700 -+++ /src/SoundTouch.h -@@ -141,7 +141,7 @@ - /// tempo/pitch/rate/samplerate settings. - #define SETTING_NOMINAL_OUTPUT_SEQUENCE 7 - --class SoundTouch : public FIFOProcessor -+class EXPORT SoundTouch : public FIFOProcessor - { - private: - /// Rate transposer class instance -@@ -160,7 +160,7 @@ - float virtualPitch; - - /// Flag: Has sample rate been set? -- BOOL bSrateSet; -+ bool bSrateSet; - - /// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and - /// 'virtualPitch' parameters. -@@ -247,8 +247,8 @@ - /// Changes a setting controlling the processing system behaviour. See the - /// 'SETTING_...' defines for available setting ID's. - /// -- /// \return 'TRUE' if the setting was succesfully changed -- BOOL setSetting(int settingId, ///< Setting ID number. see SETTING_... defines. -+ /// \return 'true' if the setting was succesfully changed -+ bool setSetting(int settingId, ///< Setting ID number. see SETTING_... defines. - int value ///< New setting value. - ); - ---- /src/RateTransposer.cpp -+++ /src/RateTransposer.cpp -@@ -120,17 +120,17 @@ RateTransposer *RateTransposer::newInsta - #endif - } - - - // Constructor - RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer) - { - numChannels = 2; -- bUseAAFilter = TRUE; -+ bUseAAFilter = true; - fRate = 0; - - // Instantiates the anti-alias filter with default tap length - // of 32 - pAAFilter = new AAFilter(32); - } - - -@@ -138,24 +138,24 @@ RateTransposer::RateTransposer() : FIFOP - RateTransposer::~RateTransposer() - { - delete pAAFilter; - } - - - - /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable --void RateTransposer::enableAAFilter(BOOL newMode) -+void RateTransposer::enableAAFilter(bool newMode) - { - bUseAAFilter = newMode; - } - - - /// Returns nonzero if anti-alias filter is enabled. --BOOL RateTransposer::isAAFilterEnabled() const -+bool RateTransposer::isAAFilterEnabled() const - { - return bUseAAFilter; - } - - - AAFilter *RateTransposer::getAAFilter() - { - return pAAFilter; -@@ -281,17 +281,17 @@ void RateTransposer::processSamples(cons - uint count; - uint sizeReq; - - if (nSamples == 0) return; - assert(pAAFilter); - - // If anti-alias filter is turned off, simply transpose without applying - // the filter -- if (bUseAAFilter == FALSE) -+ if (bUseAAFilter == false) - { - sizeReq = (uint)((float)nSamples / fRate + 1.0f); - count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples); - outputBuffer.putSamples(count); - return; - } - - // Transpose with anti-alias filter ---- /src/RateTransposer.h -+++ /src/RateTransposer.h -@@ -76,17 +76,17 @@ protected: - FIFOSampleBuffer storeBuffer; - - /// Buffer for keeping samples between transposing & anti-alias filter - FIFOSampleBuffer tempBuffer; - - /// Output sample buffer - FIFOSampleBuffer outputBuffer; - -- BOOL bUseAAFilter; -+ bool bUseAAFilter; - - virtual void resetRegisters() = 0; - - virtual uint transposeStereo(SAMPLETYPE *dest, - const SAMPLETYPE *src, - uint numSamples) = 0; - virtual uint transposeMono(SAMPLETYPE *dest, - const SAMPLETYPE *src, -@@ -126,20 +126,20 @@ public: - - /// Returns the store buffer object - FIFOSamplePipe *getStore() { return &storeBuffer; }; - - /// Return anti-alias filter object - AAFilter *getAAFilter(); - - /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable -- void enableAAFilter(BOOL newMode); -+ void enableAAFilter(bool newMode); - - /// Returns nonzero if anti-alias filter is enabled. -- BOOL isAAFilterEnabled() const; -+ bool isAAFilterEnabled() const; - - /// Sets new target rate. Normal rate = 1.0, smaller values represent slower - /// rate, larger faster rates. - virtual void setRate(float newRate); - - /// Sets the number of channels, 1 = mono, 2 = stereo - void setChannels(int channels); - ---- /src/SoundTouch.cpp -+++ /src/SoundTouch.cpp -@@ -106,17 +106,17 @@ SoundTouch::SoundTouch() - - virtualPitch = - virtualRate = - virtualTempo = 1.0; - - calcEffectiveRateAndTempo(); - - channels = 0; -- bSrateSet = FALSE; -+ bSrateSet = false; - } - - - - SoundTouch::~SoundTouch() - { - delete pRateTransposer; - delete pTDStretch; -@@ -277,27 +277,27 @@ void SoundTouch::calcEffectiveRateAndTem - } - } - } - - - // Sets sample rate. - void SoundTouch::setSampleRate(uint srate) - { -- bSrateSet = TRUE; -+ bSrateSet = true; - // set sample rate, leave other tempo changer parameters as they are. - pTDStretch->setParameters((int)srate); - } - - - // 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) -+ 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"); - } - -@@ -382,57 +382,57 @@ void SoundTouch::flush() - pTDStretch->clearInput(); - // yet leave the 'tempoChanger' 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) -+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; -+ pRateTransposer->enableAAFilter((value != 0) ? true : false); -+ return true; - - case SETTING_AA_FILTER_LENGTH : - // sets anti-alias filter length - pRateTransposer->getAAFilter()->setLength(value); -- return TRUE; -+ return true; - - case SETTING_USE_QUICKSEEK : - // enables / disables tempo routine quick seeking algorithm -- pTDStretch->enableQuickSeek((value != 0) ? TRUE : FALSE); -- return TRUE; -+ 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; -+ return true; - - case SETTING_SEEKWINDOW_MS: - // change time-stretch seek window length parameter - pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs); -- return TRUE; -+ return true; - - case SETTING_OVERLAP_MS: - // change time-stretch overlap length parameter - pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value); -- return TRUE; -+ return true; - - default : -- return FALSE; -+ return false; - } - } - - - // Reads a setting controlling the processing system behaviour. See the - // 'SETTING_...' defines for available setting ID's. - // - // Returns the setting value. ---- /src/TDStretch.cpp -+++ /src/TDStretch.cpp -@@ -81,25 +81,25 @@ static const short _scanOffsets[5][24]={ - * - * Implementation of the class 'TDStretch' - * - *****************************************************************************/ - - - TDStretch::TDStretch() : FIFOProcessor(&outputBuffer) - { -- bQuickSeek = FALSE; -+ bQuickSeek = false; - channels = 2; - - pMidBuffer = NULL; - pMidBufferUnaligned = NULL; - overlapLength = 0; - -- bAutoSeqSetting = TRUE; -- bAutoSeekSetting = TRUE; -+ bAutoSeqSetting = true; -+ bAutoSeekSetting = true; - - // outDebt = 0; - skipFract = 0; - - tempo = 1.0f; - setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS); - setTempo(1.0f); - -@@ -129,33 +129,33 @@ void TDStretch::setParameters(int aSampl - { - // accept only positive parameter values - if zero or negative, use old values instead - if (aSampleRate > 0) this->sampleRate = aSampleRate; - if (aOverlapMS > 0) this->overlapMs = aOverlapMS; - - if (aSequenceMS > 0) - { - this->sequenceMs = aSequenceMS; -- bAutoSeqSetting = FALSE; -+ bAutoSeqSetting = false; - } - else if (aSequenceMS == 0) - { - // if zero, use automatic setting -- bAutoSeqSetting = TRUE; -+ bAutoSeqSetting = true; - } - - if (aSeekWindowMS > 0) - { - this->seekWindowMs = aSeekWindowMS; -- bAutoSeekSetting = FALSE; -+ bAutoSeekSetting = false; - } - else if (aSeekWindowMS == 0) - { - // if zero, use automatic setting -- bAutoSeekSetting = TRUE; -+ bAutoSeekSetting = true; - } - - calcSeqParameters(); - - calculateOverlapLength(overlapMs); - - // set tempo to recalculate 'sampleReq' - setTempo(tempo); -@@ -229,24 +229,24 @@ void TDStretch::clear() - outputBuffer.clear(); - clearInput(); - } - - - - // Enables/disables the quick position seeking algorithm. Zero to disable, nonzero - // to enable --void TDStretch::enableQuickSeek(BOOL enable) -+void TDStretch::enableQuickSeek(bool enable) - { - bQuickSeek = enable; - } - - - // Returns nonzero if the quick seeking algorithm is enabled. --BOOL TDStretch::isQuickSeekEnabled() const -+bool TDStretch::isQuickSeekEnabled() const - { - return bQuickSeek; - } - - - // Seeks for the optimal overlap-mixing position. - int TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos) - { ---- /src/TDStretch.h -+++ /src/TDStretch.h -@@ -120,24 +120,24 @@ protected: - int seekLength; - int seekWindowLength; - int overlapDividerBits; - int slopingDivider; - float nominalSkip; - float skipFract; - FIFOSampleBuffer outputBuffer; - FIFOSampleBuffer inputBuffer; -- BOOL bQuickSeek; -+ bool bQuickSeek; - - int sampleRate; - int sequenceMs; - int seekWindowMs; - int overlapMs; -- BOOL bAutoSeqSetting; -- BOOL bAutoSeekSetting; -+ bool bAutoSeqSetting; -+ bool bAutoSeekSetting; - - void acceptNewOverlapLength(int newOverlapLength); - - virtual void clearCrossCorrState(); - void calculateOverlapLength(int overlapMs); - - virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const; - -@@ -188,20 +188,20 @@ public: - /// 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); -+ void enableQuickSeek(bool enable); - - /// Returns nonzero if the quick seeking algorithm is enabled. -- BOOL isQuickSeekEnabled() const; -+ 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 -only in patch2: ---- /src/cpu_detect_x86.cpp 2012-04-12 19:52:12.743376976 -0700 -+++ /src/cpu_detect_x86.cpp 2012-08-02 09:54:24.561712171 -0700 -@@ -38,30 +38,35 @@ - // 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(__i386__) -- // gcc -+ #if defined(__GNUC__) -+ // gcc and clang - #include "cpuid.h" - #endif - - #if defined(_M_IX86) - // windows - #include -- #define bit_MMX (1 << 23) -- #define bit_SSE (1 << 25) -- #define bit_SSE2 (1 << 26) - #endif -- -+ // If we still don't have the macros, define them (Windows, MacOS) -+ #ifndef bit_MMX -+ #define bit_MMX (1 << 23) -+ #endif -+ #ifndef bit_SSE -+ #define bit_SSE (1 << 25) -+ #endif -+ #ifndef bit_SSE2 -+ #define bit_SSE2 (1 << 26) -+ #endif - #endif - - - ////////////////////////////////////////////////////////////////////////////// - // - // processor instructions extension detection routines - // - ////////////////////////////////////////////////////////////////////////////// diff --git a/media/libsoundtouch/src/AAFilter.cpp b/media/libsoundtouch/src/AAFilter.cpp deleted file mode 100644 index 29de4096b4b6..000000000000 --- a/media/libsoundtouch/src/AAFilter.cpp +++ /dev/null @@ -1,184 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2009-01-11 03:34:24 -0800 (Sun, 11 Jan 2009) $ -// File revision : $Revision: 4 $ -// -// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#include -#include -#include -#include "AAFilter.h" -#include "FIRFilter.h" - -using namespace soundtouch; - -#define PI 3.141592655357989 -#define TWOPI (2 * PI) - -/***************************************************************************** - * - * 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 fc2, 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]; - - fc2 = 2.0 * cutoffFreq; - wc = PI * fc2; - 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 = fc2 * 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 ++) - { - // scale & round to nearest integer - temp = work[i] * scaleCoeff; - 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); - - 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); -} - - -uint AAFilter::getLength() const -{ - return pFIR->getLength(); -} diff --git a/media/libsoundtouch/src/AAFilter.h b/media/libsoundtouch/src/AAFilter.h deleted file mode 100644 index 7f172842e848..000000000000 --- a/media/libsoundtouch/src/AAFilter.h +++ /dev/null @@ -1,91 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2008-02-10 08:26:55 -0800 (Sun, 10 Feb 2008) $ -// File revision : $Revision: 4 $ -// -// $Id: AAFilter.h 11 2008-02-10 16:26:55Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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" - -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; -}; - -} - -#endif diff --git a/media/libsoundtouch/src/FIFOSampleBuffer.cpp b/media/libsoundtouch/src/FIFOSampleBuffer.cpp deleted file mode 100644 index bda169276fb0..000000000000 --- a/media/libsoundtouch/src/FIFOSampleBuffer.cpp +++ /dev/null @@ -1,274 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2012-06-13 12:29:53 -0700 (Wed, 13 Jun 2012) $ -// File revision : $Revision: 4 $ -// -// $Id: FIFOSampleBuffer.cpp 143 2012-06-13 19:29:53Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#include -#include -#include - -#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; - - assert(numChannels > 0); - 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 -// succesfully 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 enought 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 *)(((ulong)tempUnaligned + 15) & (ulong)-16); - 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; -} - diff --git a/media/libsoundtouch/src/FIFOSampleBuffer.h b/media/libsoundtouch/src/FIFOSampleBuffer.h deleted file mode 100644 index 34702d94c763..000000000000 --- a/media/libsoundtouch/src/FIFOSampleBuffer.h +++ /dev/null @@ -1,178 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2012-06-13 12:29:53 -0700 (Wed, 13 Jun 2012) $ -// File revision : $Revision: 4 $ -// -// $Id: FIFOSampleBuffer.h 143 2012-06-13 19:29:53Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 succesfully 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); - - /// 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); -}; - -} - -#endif diff --git a/media/libsoundtouch/src/FIFOSamplePipe.h b/media/libsoundtouch/src/FIFOSamplePipe.h deleted file mode 100644 index dde528c86640..000000000000 --- a/media/libsoundtouch/src/FIFOSamplePipe.h +++ /dev/null @@ -1,234 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// '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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2012-06-13 12:29:53 -0700 (Wed, 13 Jun 2012) $ -// File revision : $Revision: 4 $ -// -// $Id: FIFOSamplePipe.h 143 2012-06-13 19:29:53Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#include -#include "STTypes.h" - -namespace soundtouch -{ - -/// Abstract base class for FIFO (first-in-first-out) sample processing classes. -class FIFOSamplePipe -{ -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 deleted file mode 100644 index 3ba568f642fd..000000000000 --- a/media/libsoundtouch/src/FIRFilter.cpp +++ /dev/null @@ -1,259 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2011-09-02 11:56:11 -0700 (Fri, 02 Sep 2011) $ -// File revision : $Revision: 4 $ -// -// $Id: FIRFilter.cpp 131 2011-09-02 18:56:11Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#include -#include -#include -#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; -} - - -FIRFilter::~FIRFilter() -{ - delete[] filterCoeffs; -} - -// Usual C-version of the filter routine for stereo sound -uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const -{ - uint i, j, end; - LONG_SAMPLETYPE suml, sumr; -#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); - - end = 2 * (numSamples - length); - - for (j = 0; j < end; j += 2) - { - const SAMPLETYPE *ptr; - - suml = sumr = 0; - ptr = src + j; - - for (i = 0; i < length; i += 4) - { - // loop is unrolled by factor of 4 here for efficiency - suml += ptr[2 * i + 0] * filterCoeffs[i + 0] + - ptr[2 * i + 2] * filterCoeffs[i + 1] + - ptr[2 * i + 4] * filterCoeffs[i + 2] + - ptr[2 * i + 6] * filterCoeffs[i + 3]; - sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] + - ptr[2 * i + 3] * filterCoeffs[i + 1] + - ptr[2 * i + 5] * filterCoeffs[i + 2] + - ptr[2 * i + 7] * filterCoeffs[i + 3]; - } - -#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; -#else - suml *= dScaler; - sumr *= dScaler; -#endif // SOUNDTOUCH_INTEGER_SAMPLES - dest[j] = (SAMPLETYPE)suml; - dest[j + 1] = (SAMPLETYPE)sumr; - } - return numSamples - length; -} - - - - -// Usual C-version of the filter routine for mono sound -uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const -{ - uint i, j, end; - LONG_SAMPLETYPE sum; -#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); - - end = numSamples - length; - for (j = 0; j < end; j ++) - { - sum = 0; - for (i = 0; i < length; i += 4) - { - // loop is unrolled by factor of 4 here for efficiency - sum += src[i + 0] * filterCoeffs[i + 0] + - src[i + 1] * filterCoeffs[i + 1] + - src[i + 2] * filterCoeffs[i + 2] + - src[i + 3] * filterCoeffs[i + 3]; - } -#ifdef SOUNDTOUCH_INTEGER_SAMPLES - sum >>= resultDivFactor; - // saturate to 16 bit integer limits - sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum; -#else - sum *= dScaler; -#endif // SOUNDTOUCH_INTEGER_SAMPLES - dest[j] = (SAMPLETYPE)sum; - src ++; - } - return end; -} - - -// 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"); - - lengthDiv8 = newLength / 8; - length = lengthDiv8 * 8; - assert(length == newLength); - - resultDivFactor = uResultDivFactor; - resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor); - - delete[] filterCoeffs; - filterCoeffs = new SAMPLETYPE[length]; - memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE)); -} - - -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) const -{ - assert(numChannels == 1 || numChannels == 2); - - assert(length > 0); - assert(lengthDiv8 * 8 == length); - if (numSamples < length) return 0; - if (numChannels == 2) - { - return evaluateFilterStereo(dest, src, numSamples); - } else { - return evaluateFilterMono(dest, src, numSamples); - } -} - - - -// 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() -{ - uint uExtensions; - - uExtensions = detectCPUextensions(); - - // 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 deleted file mode 100644 index 25ede16a8aa7..000000000000 --- a/media/libsoundtouch/src/FIRFilter.h +++ /dev/null @@ -1,145 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2011-02-13 11:13:57 -0800 (Sun, 13 Feb 2011) $ -// File revision : $Revision: 4 $ -// -// $Id: FIRFilter.h 104 2011-02-13 19:13:57Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#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; - - virtual uint evaluateFilterStereo(SAMPLETYPE *dest, - const SAMPLETYPE *src, - uint numSamples) const; - virtual uint evaluateFilterMono(SAMPLETYPE *dest, - const SAMPLETYPE *src, - uint numSamples) const; - -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) const; - - 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/Makefile.in b/media/libsoundtouch/src/Makefile.in deleted file mode 100644 index 965d04318f83..000000000000 --- a/media/libsoundtouch/src/Makefile.in +++ /dev/null @@ -1,69 +0,0 @@ -# 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/. - -DEPTH = @DEPTH@ -topsrcdir = @top_srcdir@ -srcdir = @srcdir@ -VPATH = @srcdir@ - -include $(DEPTH)/config/autoconf.mk - -MODULE = soundtouch -LIBRARY_NAME = soundtouch -SHORT_LIBNAME = soundt -FORCE_SHARED_LIB = 1 -VISIBILITY_FLAGS = -EXPORTS_NAMESPACES = soundtouch - -ifeq ($(OS_ARCH),WINNT) -ifndef GNU_CC -RCFILE = $(srcdir)/soundtouch.rc -RESFILE = $(srcdir)/soundtouch.res -endif -endif - - -EXTRA_DSO_LDOPTS += $(MOZALLOC_LIB) - -# Use abort() instead of exception in SoundTouch. -DEFINES += -DST_NO_EXCEPTION_HANDLING=1 - -EXPORTS_soundtouch = SoundTouch.h \ - STTypes.h \ - FIFOSamplePipe.h \ - soundtouch_config.h \ - $(NULL) - -CPPSRCS = AAFilter.cpp \ - cpu_detect_x86.cpp \ - FIFOSampleBuffer.cpp \ - FIRFilter.cpp \ - RateTransposer.cpp \ - SoundTouch.cpp \ - TDStretch.cpp \ - $(NULL) - -ifneq (,$(INTEL_ARCHITECTURE)) -ifdef MOZ_SAMPLE_TYPE_FLOAT32 -CPPSRCS += sse_optimized.cpp -else -CPPSRCS += mmx_optimized.cpp -endif -endif - -SOUNDTOUCH_LIBS = $(DLL_PREFIX)soundtouch$(DLL_SUFFIX) - -include $(topsrcdir)/config/rules.mk - -ifneq (,$(INTEL_ARCHITECTURE)) -ifdef GNU_CC -mmx_optimized.$(OBJ_SUFFIX): CXXFLAGS+=-msse2 -sse_optimized.$(OBJ_SUFFIX): CXXFLAGS+=-msse2 -endif -ifdef SOLARIS_SUNPRO_CXX -mmx_optimized.$(OBJ_SUFFIX): OS_CXXFLAGS += -xarch=sse2 -xO4 -sse_optimized.$(OBJ_SUFFIX): OS_CXXFLAGS += -xarch=sse2 -xO4 -endif -endif - diff --git a/media/libsoundtouch/src/RateTransposer.cpp b/media/libsoundtouch/src/RateTransposer.cpp deleted file mode 100644 index 6fb8a1eaeaca..000000000000 --- a/media/libsoundtouch/src/RateTransposer.cpp +++ /dev/null @@ -1,626 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2011-09-02 11:56:11 -0700 (Fri, 02 Sep 2011) $ -// File revision : $Revision: 4 $ -// -// $Id: RateTransposer.cpp 131 2011-09-02 18:56:11Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#include -#include -#include -#include "RateTransposer.h" -#include "AAFilter.h" - -using namespace soundtouch; - - -/// A linear samplerate transposer class that uses integer arithmetics. -/// for the transposing. -class RateTransposerInteger : public RateTransposer -{ -protected: - int iSlopeCount; - int iRate; - SAMPLETYPE sPrevSampleL, sPrevSampleR; - - virtual void resetRegisters(); - - virtual uint transposeStereo(SAMPLETYPE *dest, - const SAMPLETYPE *src, - uint numSamples); - virtual uint transposeMono(SAMPLETYPE *dest, - const SAMPLETYPE *src, - uint numSamples); - -public: - RateTransposerInteger(); - virtual ~RateTransposerInteger(); - - /// Sets new target rate. Normal rate = 1.0, smaller values represent slower - /// rate, larger faster rates. - virtual void setRate(float newRate); - -}; - - -/// A linear samplerate transposer class that uses floating point arithmetics -/// for the transposing. -class RateTransposerFloat : public RateTransposer -{ -protected: - float fSlopeCount; - SAMPLETYPE sPrevSampleL, sPrevSampleR; - - virtual void resetRegisters(); - - virtual uint transposeStereo(SAMPLETYPE *dest, - const SAMPLETYPE *src, - uint numSamples); - virtual uint transposeMono(SAMPLETYPE *dest, - const SAMPLETYPE *src, - uint numSamples); - -public: - RateTransposerFloat(); - virtual ~RateTransposerFloat(); -}; - - - - -// 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 * RateTransposer::operator new(size_t s) -{ - ST_THROW_RT_ERROR("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!"); - return newInstance(); -} - - -RateTransposer *RateTransposer::newInstance() -{ -#ifdef SOUNDTOUCH_INTEGER_SAMPLES - return ::new RateTransposerInteger; -#else - return ::new RateTransposerFloat; -#endif -} - - -// Constructor -RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer) -{ - numChannels = 2; - bUseAAFilter = true; - fRate = 0; - - // Instantiates the anti-alias filter with default tap length - // of 32 - pAAFilter = new AAFilter(32); -} - - - -RateTransposer::~RateTransposer() -{ - delete pAAFilter; -} - - - -/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable -void RateTransposer::enableAAFilter(bool newMode) -{ - bUseAAFilter = newMode; -} - - -/// 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(float newRate) -{ - double fCutoff; - - fRate = newRate; - - // design a new anti-alias filter - if (newRate > 1.0f) - { - fCutoff = 0.5f / newRate; - } - else - { - fCutoff = 0.5f * newRate; - } - pAAFilter->setCutoffFreq(fCutoff); -} - - -// Outputs as many samples of the 'outputBuffer' as possible, and if there's -// any room left, outputs also as many of the incoming samples as possible. -// The goal is to drive the outputBuffer empty. -// -// It's allowed for 'output' and 'input' parameters to point to the same -// memory position. -/* -void RateTransposer::flushStoreBuffer() -{ - if (storeBuffer.isEmpty()) return; - - outputBuffer.moveSamples(storeBuffer); -} -*/ - - -// 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 up the sample rate, causing the observed playback 'rate' of the -// sound to decrease -void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples) -{ - uint count, sizeTemp, num; - - // If the parameter 'uRate' value is smaller than 'SCALE', first transpose - // the samples and then apply the anti-alias filter to remove aliasing. - - // First check that there's enough room in 'storeBuffer' - // (+16 is to reserve some slack in the destination buffer) - sizeTemp = (uint)((float)nSamples / fRate + 16.0f); - - // Transpose the samples, store the result into the end of "storeBuffer" - count = transpose(storeBuffer.ptrEnd(sizeTemp), src, nSamples); - storeBuffer.putSamples(count); - - // Apply the anti-alias filter to samples in "store output", output the - // result to "dest" - num = storeBuffer.numSamples(); - count = pAAFilter->evaluate(outputBuffer.ptrEnd(num), - storeBuffer.ptrBegin(), num, (uint)numChannels); - outputBuffer.putSamples(count); - - // Remove the processed samples from "storeBuffer" - storeBuffer.receiveSamples(count); -} - - -// Transposes down the sample rate, causing the observed playback 'rate' of the -// sound to increase -void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples) -{ - uint count, sizeTemp; - - // If the parameter 'uRate' value is larger than 'SCALE', first apply the - // anti-alias filter to remove high frequencies (prevent them from folding - // over the lover frequencies), then transpose. - - // Add the new samples to the end of the storeBuffer - storeBuffer.putSamples(src, nSamples); - - // Anti-alias filter the samples to prevent folding and output the filtered - // data to tempBuffer. Note : because of the FIR filter length, the - // filtering routine takes in 'filter_length' more samples than it outputs. - assert(tempBuffer.isEmpty()); - sizeTemp = storeBuffer.numSamples(); - - count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp), - storeBuffer.ptrBegin(), sizeTemp, (uint)numChannels); - - if (count == 0) return; - - // Remove the filtered samples from 'storeBuffer' - storeBuffer.receiveSamples(count); - - // Transpose the samples (+16 is to reserve some slack in the destination buffer) - sizeTemp = (uint)((float)nSamples / fRate + 16.0f); - count = transpose(outputBuffer.ptrEnd(sizeTemp), tempBuffer.ptrBegin(), count); - outputBuffer.putSamples(count); -} - - -// 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; - uint sizeReq; - - if (nSamples == 0) return; - assert(pAAFilter); - - // If anti-alias filter is turned off, simply transpose without applying - // the filter - if (bUseAAFilter == false) - { - sizeReq = (uint)((float)nSamples / fRate + 1.0f); - count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples); - outputBuffer.putSamples(count); - return; - } - - // Transpose with anti-alias filter - if (fRate < 1.0f) - { - upsample(src, nSamples); - } - else - { - downsample(src, nSamples); - } -} - - -// Transposes the sample rate of the given samples using linear interpolation. -// Returns the number of samples returned in the "dest" buffer -inline uint RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples) -{ - if (numChannels == 2) - { - return transposeStereo(dest, src, nSamples); - } - else - { - return transposeMono(dest, src, nSamples); - } -} - - -// Sets the number of channels, 1 = mono, 2 = stereo -void RateTransposer::setChannels(int nChannels) -{ - assert(nChannels > 0); - if (numChannels == nChannels) return; - - assert(nChannels == 1 || nChannels == 2); - numChannels = nChannels; - - storeBuffer.setChannels(numChannels); - tempBuffer.setChannels(numChannels); - outputBuffer.setChannels(numChannels); - - // Inits the linear interpolation registers - resetRegisters(); -} - - -// Clears all the samples in the object -void RateTransposer::clear() -{ - outputBuffer.clear(); - storeBuffer.clear(); -} - - -// 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 storeBuffer.isEmpty(); -} - - -////////////////////////////////////////////////////////////////////////////// -// -// RateTransposerInteger - integer arithmetic implementation -// - -/// fixed-point interpolation routine precision -#define SCALE 65536 - -// Constructor -RateTransposerInteger::RateTransposerInteger() : RateTransposer() -{ - // Notice: use local function calling syntax for sake of clarity, - // to indicate the fact that C++ constructor can't call virtual functions. - RateTransposerInteger::resetRegisters(); - RateTransposerInteger::setRate(1.0f); -} - - -RateTransposerInteger::~RateTransposerInteger() -{ -} - - -void RateTransposerInteger::resetRegisters() -{ - iSlopeCount = 0; - sPrevSampleL = - sPrevSampleR = 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 -uint RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples) -{ - unsigned int i, used; - LONG_SAMPLETYPE temp, vol1; - - if (nSamples == 0) return 0; // no samples, no work - - used = 0; - i = 0; - - // Process the last sample saved from the previous call first... - while (iSlopeCount <= SCALE) - { - vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount); - temp = vol1 * sPrevSampleL + iSlopeCount * src[0]; - dest[i] = (SAMPLETYPE)(temp / SCALE); - i++; - iSlopeCount += iRate; - } - // now always (iSlopeCount > SCALE) - iSlopeCount -= SCALE; - - while (1) - { - while (iSlopeCount > SCALE) - { - iSlopeCount -= SCALE; - used ++; - if (used >= nSamples - 1) goto end; - } - vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount); - temp = src[used] * vol1 + iSlopeCount * src[used + 1]; - dest[i] = (SAMPLETYPE)(temp / SCALE); - - i++; - iSlopeCount += iRate; - } -end: - // Store the last sample for the next round - sPrevSampleL = src[nSamples - 1]; - - 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 -uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples) -{ - unsigned int srcPos, i, used; - LONG_SAMPLETYPE temp, vol1; - - if (nSamples == 0) return 0; // no samples, no work - - used = 0; - i = 0; - - // Process the last sample saved from the sPrevSampleLious call first... - while (iSlopeCount <= SCALE) - { - vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount); - temp = vol1 * sPrevSampleL + iSlopeCount * src[0]; - dest[2 * i] = (SAMPLETYPE)(temp / SCALE); - temp = vol1 * sPrevSampleR + iSlopeCount * src[1]; - dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE); - i++; - iSlopeCount += iRate; - } - // now always (iSlopeCount > SCALE) - iSlopeCount -= SCALE; - - while (1) - { - while (iSlopeCount > SCALE) - { - iSlopeCount -= SCALE; - used ++; - if (used >= nSamples - 1) goto end; - } - srcPos = 2 * used; - vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount); - temp = src[srcPos] * vol1 + iSlopeCount * src[srcPos + 2]; - dest[2 * i] = (SAMPLETYPE)(temp / SCALE); - temp = src[srcPos + 1] * vol1 + iSlopeCount * src[srcPos + 3]; - dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE); - - i++; - iSlopeCount += iRate; - } -end: - // Store the last sample for the next round - sPrevSampleL = src[2 * nSamples - 2]; - sPrevSampleR = src[2 * nSamples - 1]; - - return i; -} - - -// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower -// iRate, larger faster iRates. -void RateTransposerInteger::setRate(float newRate) -{ - iRate = (int)(newRate * SCALE + 0.5f); - RateTransposer::setRate(newRate); -} - - -////////////////////////////////////////////////////////////////////////////// -// -// RateTransposerFloat - floating point arithmetic implementation -// -////////////////////////////////////////////////////////////////////////////// - -// Constructor -RateTransposerFloat::RateTransposerFloat() : RateTransposer() -{ - // Notice: use local function calling syntax for sake of clarity, - // to indicate the fact that C++ constructor can't call virtual functions. - RateTransposerFloat::resetRegisters(); - RateTransposerFloat::setRate(1.0f); -} - - -RateTransposerFloat::~RateTransposerFloat() -{ -} - - -void RateTransposerFloat::resetRegisters() -{ - fSlopeCount = 0; - sPrevSampleL = - sPrevSampleR = 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 -uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples) -{ - unsigned int i, used; - - used = 0; - i = 0; - - // Process the last sample saved from the previous call first... - while (fSlopeCount <= 1.0f) - { - dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]); - i++; - fSlopeCount += fRate; - } - fSlopeCount -= 1.0f; - - if (nSamples > 1) - { - while (1) - { - while (fSlopeCount > 1.0f) - { - fSlopeCount -= 1.0f; - used ++; - if (used >= nSamples - 1) goto end; - } - dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[used] + fSlopeCount * src[used + 1]); - i++; - fSlopeCount += fRate; - } - } -end: - // Store the last sample for the next round - sPrevSampleL = src[nSamples - 1]; - - 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 -uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples) -{ - unsigned int srcPos, i, used; - - if (nSamples == 0) return 0; // no samples, no work - - used = 0; - i = 0; - - // Process the last sample saved from the sPrevSampleLious call first... - while (fSlopeCount <= 1.0f) - { - dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]); - dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleR + fSlopeCount * src[1]); - i++; - fSlopeCount += fRate; - } - // now always (iSlopeCount > 1.0f) - fSlopeCount -= 1.0f; - - if (nSamples > 1) - { - while (1) - { - while (fSlopeCount > 1.0f) - { - fSlopeCount -= 1.0f; - used ++; - if (used >= nSamples - 1) goto end; - } - srcPos = 2 * used; - - dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos] - + fSlopeCount * src[srcPos + 2]); - dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos + 1] - + fSlopeCount * src[srcPos + 3]); - - i++; - fSlopeCount += fRate; - } - } -end: - // Store the last sample for the next round - sPrevSampleL = src[2 * nSamples - 2]; - sPrevSampleR = src[2 * nSamples - 1]; - - return i; -} diff --git a/media/libsoundtouch/src/RateTransposer.h b/media/libsoundtouch/src/RateTransposer.h deleted file mode 100644 index c2256210d927..000000000000 --- a/media/libsoundtouch/src/RateTransposer.h +++ /dev/null @@ -1,159 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2009-02-21 08:00:14 -0800 (Sat, 21 Feb 2009) $ -// File revision : $Revision: 4 $ -// -// $Id: RateTransposer.h 63 2009-02-21 16:00:14Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#include "AAFilter.h" -#include "FIFOSamplePipe.h" -#include "FIFOSampleBuffer.h" - -#include "STTypes.h" - -namespace soundtouch -{ - -/// A common linear samplerate transposer class. -/// -/// Note: Use function "RateTransposer::newInstance()" to create a new class -/// instance instead of the "new" operator; that function automatically -/// chooses a correct implementation depending on if integer or floating -/// arithmetics are to be used. -class RateTransposer : public FIFOProcessor -{ -protected: - /// Anti-alias filter object - AAFilter *pAAFilter; - - float fRate; - - int numChannels; - - /// Buffer for collecting samples to feed the anti-alias filter between - /// two batches - FIFOSampleBuffer storeBuffer; - - /// Buffer for keeping samples between transposing & anti-alias filter - FIFOSampleBuffer tempBuffer; - - /// Output sample buffer - FIFOSampleBuffer outputBuffer; - - bool bUseAAFilter; - - virtual void resetRegisters() = 0; - - virtual uint transposeStereo(SAMPLETYPE *dest, - const SAMPLETYPE *src, - uint numSamples) = 0; - virtual uint transposeMono(SAMPLETYPE *dest, - const SAMPLETYPE *src, - uint numSamples) = 0; - inline uint transpose(SAMPLETYPE *dest, - const SAMPLETYPE *src, - uint numSamples); - - void downsample(const SAMPLETYPE *src, - uint numSamples); - void upsample(const SAMPLETYPE *src, - uint numSamples); - - /// 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(); - - /// Operator 'new' is overloaded so that it automatically creates a suitable instance - /// depending on if we're to use integer or floating point arithmetics. - 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 implementation, depending on if - /// integer ot floating point arithmetics are to be used. - static RateTransposer *newInstance(); - - /// Returns the output buffer object - FIFOSamplePipe *getOutput() { return &outputBuffer; }; - - /// Returns the store buffer object - FIFOSamplePipe *getStore() { return &storeBuffer; }; - - /// 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(float 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; -}; - -} - -#endif diff --git a/media/libsoundtouch/src/STTypes.h b/media/libsoundtouch/src/STTypes.h deleted file mode 100644 index bfd08f2da0e3..000000000000 --- a/media/libsoundtouch/src/STTypes.h +++ /dev/null @@ -1,159 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2012-04-01 10:01:42 -0700 (Sun, 01 Apr 2012) $ -// File revision : $Revision: 3 $ -// -// $Id: STTypes.h 136 2012-04-01 17:01:42Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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; - -#include "soundtouch_config.h" - -#ifdef WIN32 -#define EXPORT __declspec(dllexport) -#else -#define EXPORT -#endif - -namespace soundtouch -{ - /// 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 !(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 take minor shortcuts - // for improved performance. Undefine to require faithfully similar SIMD - // calculations as in normal C implementation. - #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 - #define SOUNDTOUCH_ALLOW_MMX 1 - #endif - - #else - - // floating point samples - typedef float SAMPLETYPE; - // data type for sample accumulation: Use double to utilize full precision. - typedef double LONG_SAMPLETYPE; - - #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS - // Allow SSE optimizations - #define SOUNDTOUCH_ALLOW_SSE 1 - #endif - - #endif // SOUNDTOUCH_INTEGER_SAMPLES - -} - -// 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 - #define ST_THROW_RT_ERROR(x) {assert((const char *)x);} -#else - // use c++ standard exceptions - #include - #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 deleted file mode 100644 index 46de76cb2d4e..000000000000 --- a/media/libsoundtouch/src/SoundTouch.cpp +++ /dev/null @@ -1,501 +0,0 @@ -////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2012-06-13 12:29:53 -0700 (Wed, 13 Jun 2012) $ -// File revision : $Revision: 4 $ -// -// $Id: SoundTouch.cpp 143 2012-06-13 19:29:53Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#include -#include -#include -#include - -#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 = RateTransposer::newInstance(); - pTDStretch = TDStretch::newInstance(); - - setOutPipe(pTDStretch); - - rate = tempo = 0; - - virtualPitch = - virtualRate = - virtualTempo = 1.0; - - calcEffectiveRateAndTempo(); - - 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 (numChannels != 1 && numChannels != 2) - { - ST_THROW_RT_ERROR("Illegal number of channels"); - } - 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(float newRate) -{ - virtualRate = newRate; - calcEffectiveRateAndTempo(); -} - - - -// Sets new rate control value as a difference in percents compared -// to the original rate (-50 .. +100 %) -void SoundTouch::setRateChange(float newRate) -{ - virtualRate = 1.0f + 0.01f * newRate; - calcEffectiveRateAndTempo(); -} - - - -// Sets new tempo control value. Normal tempo = 1.0, smaller values -// represent slower tempo, larger faster tempo. -void SoundTouch::setTempo(float newTempo) -{ - virtualTempo = newTempo; - calcEffectiveRateAndTempo(); -} - - - -// Sets new tempo control value as a difference in percents compared -// to the original tempo (-50 .. +100 %) -void SoundTouch::setTempoChange(float newTempo) -{ - virtualTempo = 1.0f + 0.01f * newTempo; - calcEffectiveRateAndTempo(); -} - - - -// Sets new pitch control value. Original pitch = 1.0, smaller values -// represent lower pitches, larger values higher pitch. -void SoundTouch::setPitch(float newPitch) -{ - virtualPitch = newPitch; - calcEffectiveRateAndTempo(); -} - - - -// Sets pitch change in octaves compared to the original pitch -// (-1.00 .. +1.00) -void SoundTouch::setPitchOctaves(float newPitch) -{ - virtualPitch = (float)exp(0.69314718056f * newPitch); - calcEffectiveRateAndTempo(); -} - - - -// Sets pitch change in semi-tones compared to the original pitch -// (-12 .. +12) -void SoundTouch::setPitchSemiTones(int newPitch) -{ - setPitchOctaves((float)newPitch / 12.0f); -} - - - -void SoundTouch::setPitchSemiTones(float newPitch) -{ - setPitchOctaves(newPitch / 12.0f); -} - - -// Calculates 'effective' rate and tempo values from the -// nominal control values. -void SoundTouch::calcEffectiveRateAndTempo() -{ - float oldTempo = tempo; - float 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 - 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) -{ - bSrateSet = true; - // set sample rate, leave other tempo changer parameters as they are. - pTDStretch->setParameters((int)srate); -} - - -// 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"); - } - - // Transpose the rate of the new samples if necessary - /* Bypass the nominal setting - can introduce a click in sound when tempo/pitch control crosses the nominal value... - if (rate == 1.0f) - { - // The rate value is same as the original, simply evaluate the tempo changer. - assert(output == pTDStretch); - if (pRateTransposer->isEmpty() == 0) - { - // yet flush the last samples in the pitch transposer buffer - // (may happen if 'rate' changes from a non-zero value to zero) - pTDStretch->moveSamples(*pRateTransposer); - } - pTDStretch->putSamples(samples, nSamples); - } - */ -#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER - else 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 nUnprocessed; - int nOut; - SAMPLETYPE buff[64*2]; // note: allocate 2*64 to cater 64 sample frames of stereo sound - - // check how many samples still await processing, and scale - // that by tempo & rate to get expected output sample count - nUnprocessed = numUnprocessedSamples(); - nUnprocessed = (int)((double)nUnprocessed / (tempo * rate) + 0.5); - - nOut = numSamples(); // ready samples currently in buffer ... - nOut += nUnprocessed; // ... and how many we expect there to be in the end - - memset(buff, 0, 64 * 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 - // 8ksamples in any case) - for (i = 0; i < 128; i ++) - { - putSamples(buff, 64); - if ((int)numSamples() >= nOut) - { - // Enough new samples have appeared into the output! - // As samples come from processing with bigger chunks, now truncate it - // back to maximum "nOut" samples to improve duration accuracy - adjustAmountOfSamples(nOut); - - // finish - break; - } - } - - // Clear working buffers - pRateTransposer->clear(); - pTDStretch->clearInput(); - // yet leave the 'tempoChanger' 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 : - return pTDStretch->getInputSampleReq(); - - case SETTING_NOMINAL_OUTPUT_SEQUENCE : - return pTDStretch->getOutputBatchSize(); - - default : - return 0; - } -} - - -// Clears all the samples in the object's output and internal processing -// buffers. -void SoundTouch::clear() -{ - 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; -} diff --git a/media/libsoundtouch/src/SoundTouch.h b/media/libsoundtouch/src/SoundTouch.h deleted file mode 100644 index 42319ad85216..000000000000 --- a/media/libsoundtouch/src/SoundTouch.h +++ /dev/null @@ -1,277 +0,0 @@ -////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2012-04-04 12:47:28 -0700 (Wed, 04 Apr 2012) $ -// File revision : $Revision: 4 $ -// -// $Id: SoundTouch.h 141 2012-04-04 19:47:28Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 "1.7.0" - -/// SoundTouch library version id -#define SOUNDTOUCH_VERSION_ID (10700) - -// -// 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 nominal average processing sequence -/// size in samples. This value tells approcimate value how many input samples -/// SoundTouch needs to gather before it does DSP processing run for the sample batch. -/// -/// Notices: -/// - This is read-only parameter, i.e. setSetting ignores this parameter -/// - Returned value is approximate average value, exact processing batch -/// size may wary from time to time -/// - This parameter value is not constant but may 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 -/// - Returned value is approximate average value, exact processing batch -/// size may wary from time to time -/// - This parameter value is not constant but may change depending on -/// tempo/pitch/rate/samplerate settings. -#define SETTING_NOMINAL_OUTPUT_SEQUENCE 7 - -class EXPORT 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. - float virtualRate; - - /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters. - float virtualTempo; - - /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters. - float virtualPitch; - - /// Flag: Has sample rate been set? - bool bSrateSet; - - /// 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' - float rate; - - /// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch' - float 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(float newRate); - - /// Sets new tempo control value. Normal tempo = 1.0, smaller values - /// represent slower tempo, larger faster tempo. - void setTempo(float newTempo); - - /// Sets new rate control value as a difference in percents compared - /// to the original rate (-50 .. +100 %) - void setRateChange(float newRate); - - /// Sets new tempo control value as a difference in percents compared - /// to the original tempo (-50 .. +100 %) - void setTempoChange(float newTempo); - - /// Sets new pitch control value. Original pitch = 1.0, smaller values - /// represent lower pitches, larger values higher pitch. - void setPitch(float newPitch); - - /// Sets pitch change in octaves compared to the original pitch - /// (-1.00 .. +1.00) - void setPitchOctaves(float newPitch); - - /// Sets pitch change in semi-tones compared to the original pitch - /// (-12 .. +12) - void setPitchSemiTones(int newPitch); - void setPitchSemiTones(float newPitch); - - /// Sets the number of channels, 1 = mono, 2 = stereo - void setChannels(uint numChannels); - - /// Sets sample rate. - void setSampleRate(uint srate); - - /// 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. - ); - - /// 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 succesfully 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; - - - /// 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/TDStretch.cpp b/media/libsoundtouch/src/TDStretch.cpp deleted file mode 100644 index bec03ae2d6bb..000000000000 --- a/media/libsoundtouch/src/TDStretch.cpp +++ /dev/null @@ -1,808 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2012-04-01 12:49:30 -0700 (Sun, 01 Apr 2012) $ -// File revision : $Revision: 1.12 $ -// -// $Id: TDStretch.cpp 137 2012-04-01 19:49:30Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#include -#include -#include -#include - -#include "STTypes.h" -#include "cpu_detect.h" -#include "TDStretch.h" - -#include - -using namespace soundtouch; - -#define max(x, y) (((x) > (y)) ? (x) : (y)) - - -/***************************************************************************** - * - * Constant definitions - * - *****************************************************************************/ - -// Table for the hierarchical mixing position seeking algorithm -static 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; - -// outDebt = 0; - skipFract = 0; - - 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) 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, 2 * sizeof(SAMPLETYPE) * overlapLength); -} - - -void TDStretch::clearInput() -{ - inputBuffer.clear(); - clearMidBuffer(); -} - - -// 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 -{ - if (channels == 2) - { - // stereo sound - overlapStereo(pOutput, pInput + 2 * ovlPos); - } else { - // mono sound. - overlapMono(pOutput, pInput + 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, corr; - int i; - - bestCorr = FLT_MIN; - bestOffs = 0; - - // Scans for the best correlation value by testing each possible position - // over the permitted range. - for (i = 0; i < seekLength; i ++) - { - // Calculates correlation value for the mixing position corresponding - // to 'i' - corr = calcCrossCorr(refPos + channels * i, pMidBuffer); - // 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) - { - bestCorr = corr; - bestOffs = i; - } - } - // clear cross correlation routine state if necessary (is so e.g. in MMX routines). - clearCrossCorrState(); - - return bestOffs; -} - - -// 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::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos) -{ - int j; - int bestOffs; - double bestCorr, corr; - int scanCount, corrOffset, tempOffset; - - bestCorr = FLT_MIN; - bestOffs = _scanOffsets[0][0]; - corrOffset = 0; - tempOffset = 0; - - // Scans for the best correlation value using four-pass hierarchical search. - // - // The look-up table 'scans' has hierarchical position adjusting steps. - // In first pass the routine searhes for the highest correlation with - // relatively coarse steps, then rescans the neighbourhood of the highest - // correlation with better resolution and so on. - for (scanCount = 0;scanCount < 4; scanCount ++) - { - j = 0; - while (_scanOffsets[scanCount][j]) - { - tempOffset = corrOffset + _scanOffsets[scanCount][j]; - if (tempOffset >= seekLength) break; - - // Calculates correlation value for the mixing position corresponding - // to 'tempOffset' - corr = (double)calcCrossCorr(refPos + channels * tempOffset, pMidBuffer); - // heuristic rule to slightly favour values close to mid of the range - double tmp = (double)(2 * tempOffset - seekLength) / seekLength; - corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp)); - - // Checks for the highest correlation value - if (corr > bestCorr) - { - bestCorr = corr; - bestOffs = tempOffset; - } - j ++; - } - corrOffset = bestOffs; - } - // clear cross correlation routine state if necessary (is so e.g. in MMX routines). - clearCrossCorrState(); - - return bestOffs; -} - - - -/// 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 varius 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 125.0 - #define AUTOSEQ_AT_MAX 50.0 - #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 tempo - #define AUTOSEEK_AT_MIN 25.0 - #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(float 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 + 0.5f); - - // 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) -{ - assert(numChannels > 0); - if (channels == numChannels) return; - assert(numChannels == 1 || numChannels == 2); - - channels = numChannels; - inputBuffer.setChannels(channels); - outputBuffer.setChannels(channels); -} - - -// 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); -} -*/ - -#include - -// Processes as many processing frames of the samples 'inputBuffer', store -// the result into 'outputBuffer' -void TDStretch::processSamples() -{ - int ovlSkip, offset; - 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 tempo differs from the normal ('SCALE'), scan for the best overlapping - // position - 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); - - // ... then copy sequence samples from 'inputBuffer' to output: - - // length of sequence - temp = (seekWindowLength - 2 * overlapLength); - - // crosscheck that we don't have buffer overflow... - if ((int)inputBuffer.numSamples() < (offset + temp + overlapLength * 2)) - { - continue; // just in case, shouldn't really happen - } - - outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * (offset + overlapLength), (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 * 2) <= (int)inputBuffer.numSamples()); - memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + temp + overlapLength), - 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 * 2 + 16 / sizeof(SAMPLETYPE)]; - // ensure that 'pMidBuffer' is aligned to 16 byte boundary for efficiency - pMidBuffer = (SAMPLETYPE *)((((ulong)pMidBufferUnaligned) + 15) & (ulong)-16); - - 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() -{ - uint uExtensions; - - uExtensions = detectCPUextensions(); - - // 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 arithmetics 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; - } -} - -// 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 - overlapDividerBits = _getClosest2Power((sampleRate * aoverlapMs) / 1000.0) - 1; - if (overlapDividerBits > 9) overlapDividerBits = 9; - if (overlapDividerBits < 3) overlapDividerBits = 3; - newOvl = (int)pow(2.0, (int)overlapDividerBits + 1); // +1 => account for -1 above - - acceptNewOverlapLength(newOvl); - - // 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) const -{ - long corr; - long norm; - int i; - - corr = norm = 0; - // Same routine for stereo and mono. For stereo, unroll loop for better - // efficiency and gives slightly better resolution against rounding. - // For mono it same routine, just unrolls loop by factor of 4 - for (i = 0; i < channels * overlapLength; i += 4) - { - corr += (mixingPos[i] * compare[i] + - mixingPos[i + 1] * compare[i + 1] + - mixingPos[i + 2] * compare[i + 2] + - mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBits; - norm += (mixingPos[i] * mixingPos[i] + - mixingPos[i + 1] * mixingPos[i + 1] + - mixingPos[i + 2] * mixingPos[i + 2] + - mixingPos[i + 3] * mixingPos[i + 3]) >> overlapDividerBits; - } - - // Normalize result by dividing by sqrt(norm) - this step is easiest - // done using floating point operation - if (norm == 0) norm = 1; // to avoid div by zero - return (double)corr / sqrt((double)norm); -} - -#endif // SOUNDTOUCH_INTEGER_SAMPLES - -////////////////////////////////////////////////////////////////////////////// -// -// Floating point arithmetics 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; - } -} - - -/// 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); -} - - -double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare) const -{ - double corr; - double norm; - int i; - - corr = norm = 0; - // Same routine for stereo and mono. For Stereo, unroll by factor of 2. - // For mono it's same routine yet unrollsd by factor of 4. - for (i = 0; i < channels * overlapLength; i += 4) - { - corr += mixingPos[i] * compare[i] + - mixingPos[i + 1] * compare[i + 1]; - - norm += mixingPos[i] * mixingPos[i] + - mixingPos[i + 1] * mixingPos[i + 1]; - - // unroll the loop for better CPU efficiency: - corr += mixingPos[i + 2] * compare[i + 2] + - mixingPos[i + 3] * compare[i + 3]; - - norm += mixingPos[i + 2] * mixingPos[i + 2] + - mixingPos[i + 3] * mixingPos[i + 3]; - } - - if (norm < 1e-9) norm = 1.0; // to avoid div by zero - return corr / sqrt(norm); -} - -#endif // SOUNDTOUCH_FLOAT_SAMPLES diff --git a/media/libsoundtouch/src/TDStretch.h b/media/libsoundtouch/src/TDStretch.h deleted file mode 100644 index fcd0767e1f91..000000000000 --- a/media/libsoundtouch/src/TDStretch.h +++ /dev/null @@ -1,268 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2012-04-01 12:49:30 -0700 (Sun, 01 Apr 2012) $ -// File revision : $Revision: 4 $ -// -// $Id: TDStretch.h 137 2012-04-01 19:49:30Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#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; - float tempo; - - SAMPLETYPE *pMidBuffer; - SAMPLETYPE *pMidBufferUnaligned; - int overlapLength; - int seekLength; - int seekWindowLength; - int overlapDividerBits; - int slopingDivider; - float nominalSkip; - float skipFract; - FIFOSampleBuffer outputBuffer; - FIFOSampleBuffer inputBuffer; - bool bQuickSeek; - - int sampleRate; - int sequenceMs; - int seekWindowMs; - int overlapMs; - bool bAutoSeqSetting; - bool bAutoSeekSetting; - - void acceptNewOverlapLength(int newOverlapLength); - - virtual void clearCrossCorrState(); - void calculateOverlapLength(int overlapMs); - - virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const; - - virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos); - virtual int seekBestOverlapPositionQuick(const SAMPLETYPE *refPos); - int seekBestOverlapPosition(const SAMPLETYPE *refPos); - - virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const; - virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const; - - void clearMidBuffer(); - void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const; - - void calcSeqParameters(); - - /// 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(float 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; - } -}; - - - -// 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) const; - 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) const; - }; - -#endif /// SOUNDTOUCH_ALLOW_SSE - -} -#endif /// TDStretch_H diff --git a/media/libsoundtouch/src/cpu_detect.h b/media/libsoundtouch/src/cpu_detect.h deleted file mode 100644 index b55569060fc8..000000000000 --- a/media/libsoundtouch/src/cpu_detect.h +++ /dev/null @@ -1,62 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2008-02-10 08:26:55 -0800 (Sun, 10 Feb 2008) $ -// File revision : $Revision: 4 $ -// -// $Id: cpu_detect.h 11 2008-02-10 16:26:55Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 deleted file mode 100644 index d15c6bbace60..000000000000 --- a/media/libsoundtouch/src/cpu_detect_x86.cpp +++ /dev/null @@ -1,144 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2012-04-01 13:00:09 -0700 (Sun, 01 Apr 2012) $ -// File revision : $Revision: 4 $ -// -// $Id: cpu_detect_x86.cpp 138 2012-04-01 20:00:09Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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__) - // gcc and clang - #include "cpuid.h" - #endif - - #if defined(_M_IX86) - // windows - #include - #endif - // If we still don't have the macros, define them (Windows, MacOS) - #ifndef bit_MMX - #define bit_MMX (1 << 23) - #endif - #ifndef bit_SSE - #define bit_SSE (1 << 25) - #endif - #ifndef bit_SSE2 - #define bit_SSE2 (1 << 26) - #endif -#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__) - // 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 - // 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; - -#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 deleted file mode 100644 index 7cb65edd3e3b..000000000000 --- a/media/libsoundtouch/src/mmx_optimized.cpp +++ /dev/null @@ -1,317 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2012-04-01 12:49:30 -0700 (Sun, 01 Apr 2012) $ -// File revision : $Revision: 4 $ -// -// $Id: mmx_optimized.cpp 137 2012-04-01 19:49:30Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#include -#include - - -// Calculates cross correlation of two buffers -double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2) const -{ - const __m64 *pVec1, *pVec2; - __m64 shifter; - __m64 accu, normaccu; - long corr, norm; - int i; - - pVec1 = (__m64*)pV1; - pVec2 = (__m64*)pV2; - - shifter = _m_from_int(overlapDividerBits); - 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_madd_pi16(pVec1[0], pVec2[0]), - _mm_madd_pi16(pVec1[1], pVec2[1])); - temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]), - _mm_madd_pi16(pVec1[1], pVec1[1])); - accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter)); - normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter)); - - temp = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), - _mm_madd_pi16(pVec1[3], pVec2[3])); - temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]), - _mm_madd_pi16(pVec1[3], pVec1[3])); - accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter)); - normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter)); - - 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(); - - // Normalize result by dividing by sqrt(norm) - this step is easiest - // done using floating point operation - if (norm == 0) norm = 1; // to avoid div by zero - - return (double)corr / sqrt((double)norm); - // Note: Warning about the missing EMMS instruction is harmless - // as it'll be called elsewhere. -} - - - -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(overlapDividerBits + 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() -{ - 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 *)(((ulong)filterCoeffsUnalign + 15) & -16); - - // 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; -} - -#endif // SOUNDTOUCH_ALLOW_MMX diff --git a/media/libsoundtouch/src/soundtouch.rc b/media/libsoundtouch/src/soundtouch.rc deleted file mode 100644 index 313039b14bf8..000000000000 --- a/media/libsoundtouch/src/soundtouch.rc +++ /dev/null @@ -1,53 +0,0 @@ - -// 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 - -// Note: if you contain versioning information in an included -// RC script, it will be discarded -// Use module.ver to explicitly set these values - -// Do not edit this file. Changes won't affect the build. - - - - -///////////////////////////////////////////////////////////////////////////// -// -// Version -// -VS_VERSION_INFO VERSIONINFO - FILEVERSION 1,7,0,0 - PRODUCTVERSION 1,7,0,0 - FILEFLAGSMASK 0x17L -#ifdef _DEBUG - FILEFLAGS 0x1L -#else - FILEFLAGS 0x0L -#endif - FILEOS 0x4L - FILETYPE 0x2L - FILESUBTYPE 0x0L -BEGIN - BLOCK "StringFileInfo" - BEGIN - BLOCK "000004b0" - BEGIN - VALUE "Comments", "SoundTouch Library licensed for 3rd party applications subject to LGPL license v2.1. Visit http://www.surina.net/soundtouch for more information about the SoundTouch library." - VALUE "FileDescription", "SoundTouch Dynamic Link Library" - VALUE "FileVersion", "1, 7, 0, 0" - VALUE "InternalName", "SoundTouch" - VALUE "LegalCopyright", "Copyright (C) Olli Parviainen 1999-2012" - VALUE "OriginalFilename", "SoundTouch.dll" - VALUE "ProductName", " SoundTouch Dynamic Link Library" - VALUE "ProductVersion", "1, 7, 0, 0" - END - END - BLOCK "VarFileInfo" - BEGIN - VALUE "Translation", 0x0, 1200 - END -END - diff --git a/media/libsoundtouch/src/soundtouch_config.h b/media/libsoundtouch/src/soundtouch_config.h deleted file mode 100644 index 6807a1a21dcb..000000000000 --- a/media/libsoundtouch/src/soundtouch_config.h +++ /dev/null @@ -1,19 +0,0 @@ -#include "mozilla/SSE.h" - -#ifdef MOZ_SAMPLE_TYPE_FLOAT32 -#define SOUNDTOUCH_FLOAT_SAMPLES 1 -#else -#define SOUNDTOUCH_INTEGER_SAMPLES 1 -#endif - -#ifndef MOZILLA_PRESUME_SSE -#ifdef MOZ_SAMPLE_TYPE_FLOAT32 -#define SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS 1 -#endif -#endif - -#ifndef MOZILLA_PRESUME_MMX -#ifdef MOZ_SAMPLE_TYPE_S16LE -#define SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS 1 -#endif -#endif diff --git a/media/libsoundtouch/src/sse_optimized.cpp b/media/libsoundtouch/src/sse_optimized.cpp deleted file mode 100644 index ea0d9b6b0a44..000000000000 --- a/media/libsoundtouch/src/sse_optimized.cpp +++ /dev/null @@ -1,361 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2012-04-01 12:49:30 -0700 (Sun, 01 Apr 2012) $ -// File revision : $Revision: 4 $ -// -// $Id: sse_optimized.cpp 137 2012-04-01 19:49:30Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// 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 -#include - -// Calculates cross correlation of two buffers -double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2) const -{ - 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 SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION - // Little cheating allowed, return valid correlation only for - // aligned locations, meaning every second round for stereo sound. - - #define _MM_LOAD _mm_load_ps - - if (((ulong)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; - double norm = sqrt(pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]); - if (norm < 1e-9) norm = 1.0; // to avoid div by zero - - float *pvSum = (float*)&vSum; - return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / 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); - */ -} - - -////////////////////////////////////////////////////////////////////////////// -// -// 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 *)(((unsigned long)filterCoeffsUnalign + 15) & (ulong)-16); - - 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(((ulong)filterCoeffsAlign) % 16 == 0); - - // filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2' - for (j = 0; j < count; j += 2) - { - const float *pSrc; - const __m128 *pFil; - __m128 sum1, sum2; - uint i; - - pSrc = (const float*)source; // source 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(dest, _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 - )); - source += 4; - dest += 4; - } - - // 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 deleted file mode 100644 index 9940601bcd0a..000000000000 --- a/media/libsoundtouch/update.sh +++ /dev/null @@ -1,40 +0,0 @@ -# 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 -# -# 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 . -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/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 - -# Remote 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 - diff --git a/mobile/android/installer/package-manifest.in b/mobile/android/installer/package-manifest.in index 9d7ed11b58d0..7253073b50ec 100644 --- a/mobile/android/installer/package-manifest.in +++ b/mobile/android/installer/package-manifest.in @@ -40,7 +40,6 @@ @BINPATH@/@DLL_PREFIX@xpcom@DLL_SUFFIX@ @BINPATH@/@DLL_PREFIX@nspr4@DLL_SUFFIX@ @BINPATH@/@DLL_PREFIX@mozalloc@DLL_SUFFIX@ -@BINPATH@/@DLL_PREFIX@soundtouch@DLL_SUFFIX@ @BINPATH@/@DLL_PREFIX@mozglue@DLL_SUFFIX@ @BINPATH@/@DLL_PREFIX@omxplugin@DLL_SUFFIX@ @BINPATH@/@DLL_PREFIX@xul@DLL_SUFFIX@ diff --git a/mobile/xul/installer/package-manifest.in b/mobile/xul/installer/package-manifest.in index 94ea7af4bec4..8b0fd4892d84 100644 --- a/mobile/xul/installer/package-manifest.in +++ b/mobile/xul/installer/package-manifest.in @@ -51,7 +51,6 @@ @BINPATH@/@DLL_PREFIX@xpcom@DLL_SUFFIX@ @BINPATH@/@DLL_PREFIX@nspr4@DLL_SUFFIX@ @BINPATH@/@DLL_PREFIX@mozalloc@DLL_SUFFIX@ -@BINPATH@/@DLL_PREFIX@soundtouch@DLL_SUFFIX@ @BINPATH@/@DLL_PREFIX@mozglue@DLL_SUFFIX@ @BINPATH@/@DLL_PREFIX@omxplugin@DLL_SUFFIX@ #ifdef XP_MACOSX diff --git a/toolkit/content/license.html b/toolkit/content/license.html index e1888afb612e..b2b4a1b881b4 100644 --- a/toolkit/content/license.html +++ b/toolkit/content/license.html @@ -87,7 +87,6 @@
  • libffi License
  • libjingle License
  • libnestegg License
    • -
  • libsoundtouch License
  • libsrtp License
  • libunwind License
  • libyuv License
    • @@ -1968,32 +1967,6 @@ OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
    -

    libsoundtouch License

    - -

    This license applies to certain files in the directory - media/libsoundtouch/src/. -

    - -
    -The SoundTouch Library Copyright © Olli Parviainen 2001-2012
-
-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., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
-
    - -
    -

    libsrtp License

    This license applies to files in the directory diff --git a/toolkit/library/Makefile.in b/toolkit/library/Makefile.in index 677f45408064..8ddeb2cb1f6a 100644 --- a/toolkit/library/Makefile.in +++ b/toolkit/library/Makefile.in @@ -358,7 +358,6 @@ EXTRA_DSO_LDOPTS += \ $(MOZ_CAIRO_OSLIBS) \ $(MOZ_APP_EXTRA_LIBS) \ $(SQLITE_LIBS) \ - $(SOUNDTOUCH_LIBS) \ $(NULL) ifdef MOZ_NATIVE_JPEG diff --git a/toolkit/mozapps/installer/packager.mk b/toolkit/mozapps/installer/packager.mk index dd63f6de6b9c..6da907a07ef3 100644 --- a/toolkit/mozapps/installer/packager.mk +++ b/toolkit/mozapps/installer/packager.mk @@ -262,7 +262,6 @@ DIST_FILES += \ libplc4.so \ libplds4.so \ libmozsqlite3.so \ - libsoundtouch.so \ libnssutil3.so \ libnss3.so \ libssl3.so \ diff --git a/toolkit/toolkit-makefiles.sh b/toolkit/toolkit-makefiles.sh index 5ae3a99c9d97..91d25475aee8 100644 --- a/toolkit/toolkit-makefiles.sh +++ b/toolkit/toolkit-makefiles.sh @@ -1737,11 +1737,3 @@ if [ "$MOZ_SPEEX_RESAMPLER" ]; then media/libspeex_resampler/src/Makefile " fi - -if [ "$MOZ_SOUNDTOUCH" ]; then - add_makefiles " - media/libsoundtouch/Makefile - media/libsoundtouch/src/Makefile - " -fi - diff --git a/toolkit/toolkit-tiers.mk b/toolkit/toolkit-tiers.mk index aecd4f56e697..4b16d2380c1c 100644 --- a/toolkit/toolkit-tiers.mk +++ b/toolkit/toolkit-tiers.mk @@ -151,12 +151,6 @@ tier_platform_dirs += \ $(NULL) endif -ifdef MOZ_SOUNDTOUCH -tier_platform_dirs += \ - media/libsoundtouch \ - $(NULL) -endif - ifdef MOZ_CUBEB tier_platform_dirs += \ media/libcubeb \