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/* * Copyright: (C) 2000 Julius O. Smith * * 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 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 * * Julius O. Smith jos@ccrma.stanford.edu * */ /* This code was modified by Bruce Forsberg (forsberg@tns.net) to make it into a C++ class */ #ifndef _AFLIBCONVERTER_H_ #define _AFLIBCONVERTER_H_ #ifdef HAVE_CONFIG_H #include#endif #ifndef MAX #define MAX(x,y) ((x)>(y) ?(x):(y)) #endif #ifndef MIN #define MIN(x,y) ((x)<(y) ?(x):(y)) #endif #define MAX_HWORD (32767) #define MIN_HWORD (-32768) #define IBUFFSIZE 4096 /* Input buffer size */ /*! \class aflibConverter \brief Provides sample rate conversion. This class will perform audio resampling. With the constructor you can choose the type of resampling to be done. Simple linear interpolation can be done by setting linear_interpolation to be TRUE in the constructor. The other two flags are ignored if this is set. If linear_interpolation is FALSE then some form of filtering will be done. IF high_quality is FALSE then a small filter will be performed. If high_quality is TRUE then a large filter (higher quality) will be performed. For both the small and large filters another parameter can be specified, filter_interpolation. With filter_interpolation set then the filter coefficients used for both the small and large filtering will be interpolated as well. This class was designed to stream audio data. It also expects audio data as 16 bit values. Each time a new stream is started some initialization needs to be done. Thus the function initialize should be called to initialize everything. This initialize function will set the conversion factor as well as a multiplecation factor for volume. The volume only applies to the small and large filter. Since this filter uses a history of the audio data it is possible for it to vary in amplitude. This allows users to scale the data. This class will work on any number of channels. Once everything is specified then resample should be called as many times as is necessary to process all the data. The value inCount will be returned indicating how many inArray samples were actually used to produce the output. This value can be used to indicate where the next block of inArray data should start. The resample function is driven by the outCount value specified. The inArray should contain at least: outCount / factor + extra_samples. extra_samples depends on the type of filtering done. As a rule of thumb 50 should be adequate for any type of filter. */ class aflibData; class aflibConverter { public: // Available contructors and destructors aflibConverter ( bool high_quality, bool linear_interpolation, bool filter_interpolation); ~aflibConverter(); void initialize( double factor, /* factor = Sndout/Sndin */ int channels, /* number of sound channels */ double volume = 1.0); /* factor to multiply amplitude */ int resample( /* number of output samples returned */ int& inCount, /* number of input samples to convert */ int outCount, /* number of output samples to compute */ short inArray[], /* input array data (length inCount * nChans) */ short outArray[]);/* output array data (length outCount * nChans) */ private: aflibConverter(); aflibConverter(const aflibConverter& op); const aflibConverter& operator=(const aflibConverter& op); int err_ret(char *s); void deleteMemory(); int readData( int inCount, /* _total_ number of frames in input file */ short inArray[], /* input data */ short *outPtr[], /* array receiving chan samps */ int dataArraySize, /* size of these arrays */ int Xoff, /* read into input array starting at this index */ bool init_count); inline short WordToHword(int v, int scl) { short out; int llsb = (1<<(scl-1)); v += llsb; /* round */ v >>= scl; if (v>MAX_HWORD) { v = MAX_HWORD; } else if (v < MIN_HWORD) { v = MIN_HWORD; } out = (short) v; return out; }; int SrcLinear( short X[], short Y[], double factor, unsigned int *Time, unsigned short& Nx, unsigned short Nout); int SrcUp( short X[], short Y[], double factor, unsigned int *Time, unsigned short& Nx, unsigned short Nout, unsigned short Nwing, unsigned short LpScl, short Imp[], short ImpD[], bool Interp); int SrcUD( short X[], short Y[], double factor, unsigned int *Time, unsigned short& Nx, unsigned short Nout, unsigned short Nwing, unsigned short LpScl, short Imp[], short ImpD[], bool Interp); int FilterUp( short Imp[], short ImpD[], unsigned short Nwing, bool Interp, short *Xp, short Ph, short Inc); int FilterUD( short Imp[], short ImpD[], unsigned short Nwing, bool Interp, short *Xp, short Ph, short Inc, unsigned short dhb); int resampleFast( /* number of output samples returned */ int& inCount, /* number of input samples to convert */ int outCount, /* number of output samples to compute */ short inArray[], /* input array data (length inCount * nChans) */ short outArray[]);/* output array data (length outCount * nChans) */ int resampleWithFilter( /* number of output samples returned */ int& inCount, /* number of input samples to convert */ int outCount, /* number of output samples to compute */ short inArray[], /* input array data (length inCount * nChans) */ short outArray[], /* output array data (length outCount * nChans) */ short Imp[], short ImpD[], unsigned short LpScl, unsigned short Nmult, unsigned short Nwing); static short SMALL_FILTER_IMP[]; static short LARGE_FILTER_IMP[]; bool interpFilt; bool largeFilter; bool linearInterp; short ** _I; short ** _J; unsigned int _Time; double _factor; int _nChans; bool _initial; double _vol; }; #endif
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