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https://github.com/azahar-emu/soundtouch
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Removed commented code, style cleanup
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oparviainen
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README.html
23
README.html
@ -14,7 +14,7 @@
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<body class="normal">
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<hr>
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<h1>SoundTouch audio processing library v2.1pre</h1>
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<p class="normal">SoundTouch library Copyright © Olli Parviainen 2001-2018</p>
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<p class="normal">SoundTouch library Copyright <EFBFBD> Olli Parviainen 2001-2018</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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@ -575,16 +575,17 @@ this corresponds to lowering the pitch by -0.318 semitones:</p>
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<h3>5.1. SoundTouch library Change History </h3>
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<p><b>2.1pre:</b></p>
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<ul>
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<li>Refactored C# interface example</li>
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<li>Disable anti-alias filter when switch
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SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER defined because anti-alias
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filter cause slight click if the rate change crosses zero during
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processing</li>
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<li>Added script for building SoundTouchDll dynamic-link-library for GNU platforms</li>
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<li>Rewrote Beats-per-Minute analysis algorithm for more reliable BPM
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detection</li>
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<li>Added BPM functions to SoundTouchDll API</li>
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<li>Migrated Visual Studio project files to MSVC 201x format</li>
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<li>Refactored C# interface example</li>
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<li>Disable anti-alias filter when switch
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SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER defined because anti-alias
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filter cause slight click if the rate change crosses zero during
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processing</li>
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<li>Added script for building SoundTouchDll dynamic-link-library for GNU platforms</li>
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<li>Rewrote Beats-per-Minute analysis algorithm for more reliable BPM detection</li>
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<li>Added BPM functions to SoundTouchDll API</li>
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<li>Migrated Visual Studio project files to MSVC 201x format</li>
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<li>Replaced function parameter value asserts with runtime exceptions</li>
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<li>Code maintenance & style cleanup</li>
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</ul>
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<p><b>2.0:</b></p>
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<ul>
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@ -66,82 +66,8 @@ namespace soundtouch
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/// Maximum allowed BPM rate range. Used to restrict accepted result below a reasonable limit.
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#define MAX_BPM_VALID 190
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////////////////////////////////////////////////////////////////////////////////
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/*
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class BeatCollection
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{
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private:
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int size;
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// Ensure there's enough capacity in arrays
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void EnsureCapacity(int newCapacity)
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{
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if (newCapacity > size)
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{
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// enlarge arrays
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int oldSize = size;
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float *beatPosOld = beatPos;
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float *beatValuesOld = beatValues;
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while (size < newCapacity) size *= 2;
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printf("Alloc more %d\n", size);
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beatPos = new float[size];
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beatValues = new float[size];
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if ((beatPos == NULL) || (beatValues == NULL))
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{
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ST_THROW_RT_ERROR("can't allocate memory");
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}
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// copy old arrays to new arrays
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memcpy(beatPos, beatPosOld, sizeof(float)*oldSize);
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memcpy(beatValues, beatValuesOld, sizeof(float)*oldSize);
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// free old arrays
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delete[] beatPosOld;
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delete[] beatValuesOld;
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}
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}
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public:
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// beat position array
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float *beatPos;
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// beat values array
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float *beatValues;
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// number of beats in arrays
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int numBeats;
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// constructor
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BeatCollection()
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{
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numBeats = 0;
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size = 1024;
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beatPos = new float[size];
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beatValues = new float[size];
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}
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// destructor
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~BeatCollection()
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{
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delete[] beatPos;
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delete[] beatValues;
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}
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// add new beat position into array
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void Add(float pos, float value)
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{
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EnsureCapacity(numBeats + 1);
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beatPos[numBeats] = pos;
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beatValues[numBeats] = value;
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numBeats++;
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}
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};
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*/
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typedef struct
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{
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float pos;
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@ -257,7 +183,6 @@ public:
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int numSamples ///< Number of samples in buffer
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);
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/// Analyzes the results and returns the BPM rate. Use this function to read result
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/// after whole song data has been input to the class by consecutive calls of
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/// 'inputSamples' function.
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@ -265,7 +190,6 @@ public:
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/// \return Beats-per-minute rate, or zero if detection failed.
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float getBpm();
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/// Get beat position arrays. Note: The array includes also really low beat detection values
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/// in absence of clear strong beats. Consumer may wish to filter low values away.
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/// - "pos" receive array of beat positions
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@ -127,7 +127,6 @@ public:
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};
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/// Base-class for sound processing routines working in FIFO principle. With this base
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/// class it's easy to implement sound processing stages that can be chained together,
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/// so that samples that are fed into beginning of the pipe automatically go through
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@ -150,7 +149,6 @@ protected:
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output = pOutput;
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}
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/// Constructor. Doesn't define output pipe; it has to be set be
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/// 'setOutPipe' function.
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FIFOProcessor()
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@ -158,7 +156,6 @@ protected:
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output = NULL;
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}
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/// Constructor. Configures output pipe.
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FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
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)
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@ -166,13 +163,11 @@ protected:
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output = pOutput;
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}
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/// Destructor.
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virtual ~FIFOProcessor()
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{
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}
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/// Returns a pointer to the beginning of the output samples.
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/// This function is provided for accessing the output samples directly.
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/// Please be careful for not to corrupt the book-keeping!
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@ -199,7 +194,6 @@ public:
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return output->receiveSamples(outBuffer, maxSamples);
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}
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/// Adjusts book-keeping so that given number of samples are removed from beginning of the
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/// sample buffer without copying them anywhere.
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///
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@ -211,14 +205,12 @@ public:
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return output->receiveSamples(maxSamples);
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}
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/// Returns number of samples currently available.
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virtual uint numSamples() const
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{
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return output->numSamples();
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}
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/// Returns nonzero if there aren't any samples available for outputting.
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virtual int isEmpty() const
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{
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@ -231,7 +223,6 @@ public:
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{
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return output->adjustAmountOfSamples(numSamples);
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}
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};
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}
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@ -151,7 +151,6 @@ RunParameters::RunParameters(const int nParams, const char * const paramStr[])
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}
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// Checks parameter limits
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void RunParameters::checkLimits()
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{
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@ -184,7 +183,6 @@ void RunParameters::checkLimits()
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}
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// Unknown switch parameter -- throws an exception with an error message
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void RunParameters::throwIllegalParamExp(const string &str) const
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{
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@ -196,7 +194,6 @@ void RunParameters::throwIllegalParamExp(const string &str) const
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}
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void RunParameters::throwLicense() const
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{
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ST_THROW_RT_ERROR(licenseText);
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@ -56,7 +56,6 @@ static const char fmtStr[] = "fmt ";
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static const char factStr[] = "fact";
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static const char dataStr[] = "data";
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//////////////////////////////////////////////////////////////////////////////
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//
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// Helper functions for swapping byte order to correctly read/write WAV files
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@ -228,19 +227,10 @@ void WavInFile::init()
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ST_THROW_RT_ERROR("Error: Illegal wav file header format parameters.");
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}
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/* Ignore 'fixed' field value as 32bit signed linear data can have other value than 1.
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if (header.format.fixed != 1)
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{
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string msg = "Input file uses unsupported encoding.";
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ST_THROW_RT_ERROR(msg.c_str());
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}
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*/
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dataRead = 0;
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}
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WavInFile::~WavInFile()
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{
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if (fptr) fclose(fptr);
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@ -248,7 +238,6 @@ WavInFile::~WavInFile()
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}
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void WavInFile::rewind()
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{
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int hdrsOk;
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@ -465,7 +454,6 @@ int WavInFile::eof() const
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}
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// test if character code is between a white space ' ' and little 'z'
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static int isAlpha(char c)
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{
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@ -506,8 +494,6 @@ int WavInFile::readRIFFBlock()
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}
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int WavInFile::readHeaderBlock()
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{
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char label[5];
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@ -679,7 +665,6 @@ uint WavInFile::getSampleRate() const
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}
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uint WavInFile::getDataSizeInBytes() const
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{
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return header.data.data_len;
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@ -713,7 +698,6 @@ uint WavInFile::getElapsedMS() const
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}
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//////////////////////////////////////////////////////////////////////////////
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//
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// Class WavOutFile
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@ -752,7 +736,6 @@ WavOutFile::WavOutFile(FILE *file, int sampleRate, int bits, int channels)
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}
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WavOutFile::~WavOutFile()
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{
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finishHeader();
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@ -761,7 +744,6 @@ WavOutFile::~WavOutFile()
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}
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void WavOutFile::fillInHeader(uint sampleRate, uint bits, uint channels)
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{
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// fill in the 'riff' part..
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@ -812,7 +794,6 @@ void WavOutFile::finishHeader()
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}
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void WavOutFile::writeHeader()
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{
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WavHeader hdrTemp;
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@ -845,7 +826,6 @@ void WavOutFile::writeHeader()
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}
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void WavOutFile::write(const unsigned char *buffer, int numElems)
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{
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int res;
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@ -866,7 +846,6 @@ void WavOutFile::write(const unsigned char *buffer, int numElems)
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}
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void WavOutFile::write(const short *buffer, int numElems)
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{
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int res;
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@ -218,7 +218,6 @@ public:
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};
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/// Class for writing WAV audio files.
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class WavOutFile : protected WavFileBase
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{
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@ -108,7 +108,6 @@ static void openFiles(WavInFile **inFile, WavOutFile **outFile, const RunParamet
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}
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// Sets the 'SoundTouch' object up according to input file sound format &
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// command line parameters
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static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunParameters *params)
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@ -165,7 +164,6 @@ static void setup(SoundTouch *pSoundTouch, const WavInFile *inFile, const RunPar
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}
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// Processes the sound
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static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outFile)
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{
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@ -218,7 +216,6 @@ static void process(SoundTouch *pSoundTouch, WavInFile *inFile, WavOutFile *outF
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}
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// Detect BPM rate of inFile and adjust tempo setting accordingly if necessary
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static void detectBPM(WavInFile *inFile, RunParameters *params)
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{
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@ -274,7 +271,6 @@ static void detectBPM(WavInFile *inFile, RunParameters *params)
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}
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int main(const int nParams, const char * const paramStr[])
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{
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WavInFile *inFile;
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@ -68,7 +68,6 @@ using namespace soundtouch;
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#define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
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#endif
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/*****************************************************************************
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*
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* Implementation of the class 'AAFilter'
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@ -83,14 +82,12 @@ AAFilter::AAFilter(uint len)
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}
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AAFilter::~AAFilter()
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{
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delete pFIR;
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}
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// Sets new anti-alias filter cut-off edge frequency, scaled to
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// sampling frequency (nyquist frequency = 0.5).
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// The filter will cut frequencies higher than the given frequency.
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@ -101,7 +98,6 @@ void AAFilter::setCutoffFreq(double newCutoffFreq)
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}
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// Sets number of FIR filter taps
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void AAFilter::setLength(uint newLength)
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{
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@ -110,7 +106,6 @@ void AAFilter::setLength(uint newLength)
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}
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// Calculates coefficients for a low-pass FIR filter using Hamming window
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void AAFilter::calculateCoeffs()
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{
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@ -170,12 +165,10 @@ void AAFilter::calculateCoeffs()
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for (i = 0; i < length; i ++)
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{
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temp = work[i] * scaleCoeff;
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//#if SOUNDTOUCH_INTEGER_SAMPLES
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// scale & round to nearest integer
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temp += (temp >= 0) ? 0.5 : -0.5;
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// ensure no overfloods
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assert(temp >= -32768 && temp <= 32767);
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//#endif
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coeffs[i] = (SAMPLETYPE)temp;
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}
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@ -265,4 +265,3 @@ uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
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}
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return samplesInBuffer;
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}
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@ -68,6 +68,7 @@ FIRFilter::~FIRFilter()
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delete[] filterCoeffs;
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}
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// Usual C-version of the filter routine for stereo sound
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uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
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{
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@ -126,8 +127,6 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
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}
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// Usual C-version of the filter routine for mono sound
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uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
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{
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@ -253,7 +252,6 @@ uint FIRFilter::getLength() const
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}
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// Applies the filter to the given sequence of samples.
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//
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// Note : The amount of outputted samples is by value of 'filter_length'
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@ -283,7 +281,6 @@ uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSample
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}
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// Operator 'new' is overloaded so that it automatically creates a suitable instance
|
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// depending on if we've a MMX-capable CPU available or not.
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void * FIRFilter::operator new(size_t s)
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@ -57,7 +57,7 @@ int PeakFinder::findTop(const float *data, int peakpos) const
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refvalue = data[peakpos];
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|
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// seek within ±10 points
|
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// seek within <EFBFBD>10 points
|
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start = peakpos - 10;
|
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if (start < minPos) start = minPos;
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end = peakpos + 10;
|
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@ -171,7 +171,6 @@ double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos)
|
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}
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||||
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||||
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||||
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/// get exact center of peak near given position by calculating local mass of center
|
||||
double PeakFinder::getPeakCenter(const float *data, int peakpos) const
|
||||
{
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@ -211,7 +210,6 @@ double PeakFinder::getPeakCenter(const float *data, int peakpos) const
|
||||
}
|
||||
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||||
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||||
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double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
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{
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@ -258,7 +256,7 @@ double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
|
||||
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// accept harmonic peak if
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// (a) it is found
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// (b) is within ±4% of the expected harmonic interval
|
||||
// (b) is within <EFBFBD>4% of the expected harmonic interval
|
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// (c) has at least half x-corr value of the max. peak
|
||||
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||||
double diff = harmonic * peaktmp / highPeak;
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|
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@ -64,7 +64,6 @@ RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
|
||||
}
|
||||
|
||||
|
||||
|
||||
RateTransposer::~RateTransposer()
|
||||
{
|
||||
delete pAAFilter;
|
||||
@ -72,7 +71,6 @@ RateTransposer::~RateTransposer()
|
||||
}
|
||||
|
||||
|
||||
|
||||
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
|
||||
void RateTransposer::enableAAFilter(bool newMode)
|
||||
{
|
||||
@ -96,7 +94,6 @@ AAFilter *RateTransposer::getAAFilter()
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
|
||||
// iRate, larger faster iRates.
|
||||
void RateTransposer::setRate(double newRate)
|
||||
|
||||
@ -125,21 +125,9 @@ 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 arithmetic.
|
||||
// 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 or floating point arithmetic 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();
|
||||
|
||||
|
||||
@ -111,7 +111,6 @@ SoundTouch::SoundTouch()
|
||||
}
|
||||
|
||||
|
||||
|
||||
SoundTouch::~SoundTouch()
|
||||
{
|
||||
delete pRateTransposer;
|
||||
@ -119,7 +118,6 @@ SoundTouch::~SoundTouch()
|
||||
}
|
||||
|
||||
|
||||
|
||||
/// Get SoundTouch library version string
|
||||
const char *SoundTouch::getVersionString()
|
||||
{
|
||||
@ -156,7 +154,6 @@ void SoundTouch::setRate(double newRate)
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Sets new rate control value as a difference in percents compared
|
||||
// to the original rate (-50 .. +100 %)
|
||||
void SoundTouch::setRateChange(double newRate)
|
||||
@ -166,7 +163,6 @@ void SoundTouch::setRateChange(double newRate)
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Sets new tempo control value. Normal tempo = 1.0, smaller values
|
||||
// represent slower tempo, larger faster tempo.
|
||||
void SoundTouch::setTempo(double newTempo)
|
||||
@ -176,7 +172,6 @@ void SoundTouch::setTempo(double newTempo)
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Sets new tempo control value as a difference in percents compared
|
||||
// to the original tempo (-50 .. +100 %)
|
||||
void SoundTouch::setTempoChange(double newTempo)
|
||||
@ -186,7 +181,6 @@ void SoundTouch::setTempoChange(double newTempo)
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Sets new pitch control value. Original pitch = 1.0, smaller values
|
||||
// represent lower pitches, larger values higher pitch.
|
||||
void SoundTouch::setPitch(double newPitch)
|
||||
@ -196,7 +190,6 @@ void SoundTouch::setPitch(double newPitch)
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Sets pitch change in octaves compared to the original pitch
|
||||
// (-1.00 .. +1.00)
|
||||
void SoundTouch::setPitchOctaves(double newPitch)
|
||||
@ -206,7 +199,6 @@ void SoundTouch::setPitchOctaves(double newPitch)
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Sets pitch change in semi-tones compared to the original pitch
|
||||
// (-12 .. +12)
|
||||
void SoundTouch::setPitchSemiTones(int newPitch)
|
||||
@ -215,7 +207,6 @@ void SoundTouch::setPitchSemiTones(int newPitch)
|
||||
}
|
||||
|
||||
|
||||
|
||||
void SoundTouch::setPitchSemiTones(double newPitch)
|
||||
{
|
||||
setPitchOctaves(newPitch / 12.0);
|
||||
@ -294,22 +285,6 @@ void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
|
||||
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);
|
||||
}
|
||||
*/
|
||||
|
||||
// accumulate how many samples are expected out from processing, given the current
|
||||
// processing setting
|
||||
samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo);
|
||||
@ -365,7 +340,6 @@ void SoundTouch::flush()
|
||||
delete[] buff;
|
||||
|
||||
// Clear input buffers
|
||||
// pRateTransposer->clearInput();
|
||||
pTDStretch->clearInput();
|
||||
// yet leave the output intouched as that's where the
|
||||
// flushed samples are!
|
||||
@ -502,7 +476,6 @@ int SoundTouch::getSetting(int settingId) const
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Clears all the samples in the object's output and internal processing
|
||||
// buffers.
|
||||
void SoundTouch::clear()
|
||||
@ -514,7 +487,6 @@ void SoundTouch::clear()
|
||||
}
|
||||
|
||||
|
||||
|
||||
/// Returns number of samples currently unprocessed.
|
||||
uint SoundTouch::numUnprocessedSamples() const
|
||||
{
|
||||
@ -531,7 +503,6 @@ uint SoundTouch::numUnprocessedSamples() const
|
||||
}
|
||||
|
||||
|
||||
|
||||
/// 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.
|
||||
|
||||
@ -157,7 +157,6 @@ protected:
|
||||
void calcSeqParameters();
|
||||
void adaptNormalizer();
|
||||
|
||||
|
||||
/// Changes the tempo of the given sound samples.
|
||||
/// Returns amount of samples returned in the "output" buffer.
|
||||
/// The maximum amount of samples that can be returned at a time is set by
|
||||
@ -242,7 +241,6 @@ public:
|
||||
return seekWindowLength - overlapLength;
|
||||
}
|
||||
|
||||
|
||||
/// return approximate initial input-output latency
|
||||
int getLatency() const
|
||||
{
|
||||
@ -251,7 +249,6 @@ public:
|
||||
};
|
||||
|
||||
|
||||
|
||||
// Implementation-specific class declarations:
|
||||
|
||||
#ifdef SOUNDTOUCH_ALLOW_MMX
|
||||
|
||||
@ -68,7 +68,6 @@ void disableExtensions(uint dwDisableMask)
|
||||
}
|
||||
|
||||
|
||||
|
||||
/// Checks which instruction set extensions are supported by the CPU.
|
||||
uint detectCPUextensions(void)
|
||||
{
|
||||
|
||||
@ -217,7 +217,6 @@ void TDStretchMMX::clearCrossCorrState()
|
||||
}
|
||||
|
||||
|
||||
|
||||
// MMX-optimized version of the function overlapStereo
|
||||
void TDStretchMMX::overlapStereo(short *output, const short *input) const
|
||||
{
|
||||
@ -333,7 +332,6 @@ void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uRe
|
||||
}
|
||||
|
||||
|
||||
|
||||
// mmx-optimized version of the filter routine for stereo sound
|
||||
uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
|
||||
{
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user