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3rdparty: Soundtouch bump to v2.4.0
Author also moved to https://codeberg.org/soundtouch/soundtouch. Audiochannel bump from 16 to 32 limit.
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3rdparty/soundtouch/README.html
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32
3rdparty/soundtouch/README.html
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@ -15,8 +15,8 @@
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<body class="normal">
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<hr>
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<h1>SoundTouch audio processing library v2.3.3</h1>
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<p class="normal">SoundTouch library Copyright © Olli Parviainen 2001-2024</p>
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<h1>SoundTouch audio processing library v2.4.0</h1>
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<p class="normal">SoundTouch library Copyright © Olli Parviainen 2001-2025</p>
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<hr>
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<h2>1. Introduction </h2>
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<p>SoundTouch is an open-source audio processing library that allows
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@ -81,8 +81,8 @@
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<p>The SoundTouch library compiles in practically any platform
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supporting GNU compiler (GCC) tools.
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<h4>2.2.1 Compiling with autotools</h4>
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<p>To install build prerequisites for 'autotools' tool chain:</p>
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<pre> sudo apt-get install automake autoconf libtool build-essential</pre>
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<p>To install build prerequisites for 'autotools' tool chain (for Ubuntu/Debian. Use dnf/yum/etc in other distros):</p>
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<pre> sudo apt install -y automake autoconf libtool build-essential</pre>
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<p>To build and install the binaries, run the following commands in
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/soundtouch directory:</p>
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<table border="0" cellpadding="0" cellspacing="4">
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@ -140,8 +140,8 @@
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<h4><b>2.2.2 Compiling with cmake</b></h4>
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<p>'cmake' build scripts are provided as an alternative to the autotools toolchain.</p>
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<p>To install cmake build prerequisites:</p>
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<pre> sudo apt-get install libtool build-essential cmake</pre>
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<p>To install cmake build prerequisites (for Ubuntu/Debian. Use dnf/yum/etc in other distros):</p>
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<pre> sudo apt install -y libtool build-essential cmake</pre>
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<p>To build:</p>
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<pre>
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cmake .
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@ -205,7 +205,7 @@
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separate mono channels, this isn't recommended because processing the
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channels separately would result in losing the phase coherency between
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the channels, which consequently would ruin the stereo effect.</p>
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<p>Sample rates between 8000-48000H are supported.</p>
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<p>Sample rates between 8000-48000Hz are supported.</p>
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<h3>3.2. Processing latency</h3>
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<p>The processing and latency constraints of the SoundTouch library are:</p>
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<ul>
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@ -606,10 +606,17 @@
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<pre>soundstretch original.wav output.wav -pitch=-0.318</pre>
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</blockquote>
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<hr>
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<h2>5. Change History</h2>
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<a name="changehistory"><h2>5. Change History</h2></a>
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<h3>5.1. SoundTouch library Change History </h3>
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<p><b>2.3.3:</b></p>
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<p><b>2.4.0:</b></p>
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<ul class="current">
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<li>Set CMake minimum version to 3.5 to avoid deprecation warning</li>
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<li>Increase max nr. of channels from 16 to 32</li>
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<li>Don't use `pow()` when using integer precision samples</li>
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<li>Replace `-Ofast` that's being deprecated in some compilers, by `-O3 -ffast-math`</li>
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</ul>
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<p><b>2.3.3:</b></p>
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<ul>
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<li>Fixing compiler warnings, maintenance fixes to make/build files for various systems
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</li>
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</ul>
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@ -881,9 +888,14 @@
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<ul>
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<li> Initial release</li>
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</ul>
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<br><br>
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<h3>5.2. SoundStretch application Change History </h3>
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<p><b>2.3.3:</b></p>
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<p><b>2.4.0:</b></p>
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<ul class="current_soundstretch">
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<li>parse command-line argument values with double float precision.</li>
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</ul>
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<p><b>2.3.3:</b></p>
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<ul>
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<li>Added support for Asian / non-latin filenames in Windows. Gnu platform has supported them already earlier.</li>
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</ul>
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<p><b>1.9:</b></p>
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5
3rdparty/soundtouch/soundtouch/STTypes.h
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5
3rdparty/soundtouch/soundtouch/STTypes.h
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@ -56,8 +56,9 @@ typedef unsigned long ulong;
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namespace soundtouch
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{
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/// Max allowed number of channels
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#define SOUNDTOUCH_MAX_CHANNELS 16
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/// Max allowed number of channels. This is not a hard limit but to have some
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/// maximum value for argument sanity checks -- can be increased if necessary
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#define SOUNDTOUCH_MAX_CHANNELS 32
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/// Activate these undef's to overrule the possible sampletype
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/// setting inherited from some other header file:
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4
3rdparty/soundtouch/soundtouch/SoundTouch.h
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4
3rdparty/soundtouch/soundtouch/SoundTouch.h
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@ -72,10 +72,10 @@ namespace soundtouch
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{
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/// Soundtouch library version string
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#define SOUNDTOUCH_VERSION "2.3.3"
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#define SOUNDTOUCH_VERSION "2.4.0"
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/// SoundTouch library version id
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#define SOUNDTOUCH_VERSION_ID (20303)
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#define SOUNDTOUCH_VERSION_ID (20400)
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//
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// Available setting IDs for the 'setSetting' & 'get_setting' functions:
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@ -301,7 +301,7 @@ void BPMDetect::updateXCorr(int process_samples)
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pBuffer = buffer->ptrBegin();
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// calculate decay factor for xcorr filtering
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float xcorr_decay = (float)pow(0.5, 1.0 / (XCORR_DECAY_TIME_CONSTANT * TARGET_SRATE / process_samples));
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float xcorr_decay = (float)pow(0.5, process_samples / (XCORR_DECAY_TIME_CONSTANT * TARGET_SRATE));
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// prescale pbuffer
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float tmp[XCORR_UPDATE_SEQUENCE];
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@ -56,7 +56,6 @@ using namespace soundtouch;
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FIRFilter::FIRFilter()
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{
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resultDivFactor = 0;
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resultDivider = 0;
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length = 0;
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lengthDiv8 = 0;
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filterCoeffs = nullptr;
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@ -79,7 +78,7 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
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uint ilength = length & -8;
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assert((length != 0) && (length == ilength) && (src != nullptr) && (dest != nullptr) && (filterCoeffs != nullptr));
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assert(numSamples > ilength);
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assert(numSamples >= ilength);
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end = 2 * (numSamples - ilength);
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@ -155,7 +154,7 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
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assert(src != nullptr);
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assert(dest != nullptr);
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assert(filterCoeffs != nullptr);
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assert(numChannels < 16);
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assert(numChannels <= SOUNDTOUCH_MAX_CHANNELS);
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// hint compiler autovectorization that loop length is divisible by 8
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int ilength = length & -8;
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@ -207,24 +206,24 @@ void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint u
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assert(newLength > 0);
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if (newLength % 8) ST_THROW_RT_ERROR("FIR filter length not divisible by 8");
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#ifdef SOUNDTOUCH_FLOAT_SAMPLES
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// scale coefficients already here if using floating samples
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double scale = 1.0 / resultDivider;
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#else
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short scale = 1;
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#endif
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lengthDiv8 = newLength / 8;
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length = lengthDiv8 * 8;
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assert(length == newLength);
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resultDivFactor = uResultDivFactor;
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resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor);
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delete[] filterCoeffs;
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filterCoeffs = new SAMPLETYPE[length];
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delete[] filterCoeffsStereo;
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filterCoeffsStereo = new SAMPLETYPE[length*2];
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#ifdef SOUNDTOUCH_FLOAT_SAMPLES
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// scale coefficients already here if using floating samples
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const double scale = ::pow(0.5, (int)resultDivFactor);;
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#else
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const short scale = 1;
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#endif
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for (uint i = 0; i < length; i ++)
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{
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filterCoeffs[i] = (SAMPLETYPE)(coeffs[i] * scale);
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@ -52,9 +52,6 @@ protected:
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// Result divider factor in 2^k format
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uint resultDivFactor;
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// Result divider value.
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SAMPLETYPE resultDivider;
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// Memory for filter coefficients
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SAMPLETYPE *filterCoeffs;
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SAMPLETYPE *filterCoeffsStereo;
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@ -211,9 +211,6 @@ FIRFilterSSE::~FIRFilterSSE()
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// (overloaded) Calculates filter coefficients for SSE routine
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void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
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{
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uint i;
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float fDivider;
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FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
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// Scale the filter coefficients so that it won't be necessary to scale the filtering result
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@ -223,13 +220,13 @@ void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uRe
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filterCoeffsUnalign = new float[2 * newLength + 4];
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filterCoeffsAlign = (float *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
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fDivider = (float)resultDivider;
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const float scale = ::pow(0.5, (int)resultDivFactor);
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// rearrange the filter coefficients for mmx routines
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for (i = 0; i < newLength; i ++)
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// rearrange the filter coefficients for sse routines
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for (auto i = 0U; i < newLength; i ++)
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{
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filterCoeffsAlign[2 * i + 0] =
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filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
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filterCoeffsAlign[2 * i + 1] = coeffs[i] * scale;
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}
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}
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