SndLib
Bill Schottstaedt (bil@ccrma.stanford.edu)
related documentation: snd.html grfsnd.html extsnd.html sndscm.html sndclm.html libxm.html s7.html index.html
Contents

- Introduction
- Headers
- Data
- Hardware
- Music V
- Examples
  - SndInfo
  - SndPlay
  - SndSine
  - clmosc
  - Other Examples
- Extension Languages

Introduction

sndlib is a collection of sound file and audio hardware handlers written in C and running currently in various Unices via OSS or ALSA, Mac OSX, and on old Windows systems. It provides relatively straightforward access to many sound file headers and data types, and most of the features of the audio hardware.

To build sndlib (sndlib.so if possible, and sndlib.a):

      ./configure
      make

To install it, 'make install' — I've tested this process in Linux and on a Mac.

The following files make up sndlib:

- io.c (read and write sound file data)
- headers.c (read and write sound file headers)
- audio.c (read and write sound hardware ports)
- sound.c (provide slightly higher level access to the preceding files)
- sndlib.h (header for the preceding files)
- sndlib2xen.c and sndlib-strings.h (tie preceding into s7, Ruby, or Forth)
- clm.c and clm.h (Music V implementation)
- clm2xen.c, vct.c and vct.h (tie clm.c into s7, Ruby, or Forth)
- xen.h, xen.c, xen.html (the embedded language support)
- run.c (the run macro to speed up scheme computations)

The naming scheme is more as less as follows: the sndlib prefix is "mus" so function names start with "mus_" and constants start with "MUS_". Audio hardware constants start with "MUS_AUDIO_", functions involving sound files referenced through the file name start with "mus_sound_", functions involving files at a lower level with "mus_file_", functions involving header access with "mus_header_", functions involving audio hardware access with "mus_audio_", and various others just with "mus_" (number translations, etc). Conversions use the word "to" as in "mus_samples_to_bytes". Booleans use "_p" (an ancient Common Lisp convention meaning "predicate" I think).

Headers

Sound files have built-in descriptors known as headers. The following functions return the information in the header. In each case the argument to the function is the full file name of the sound file.

      mus_long_t mus_sound_samples (const char *arg)        /* samples of sound according to header */
      mus_long_t mus_sound_frames (const char *arg)         /* samples per channel */
      float mus_sound_duration (const char *arg)       /* sound duration in seconds */
      mus_long_t mus_sound_length (const char *arg)         /* true file length in bytes */

      int mus_sound_datum_size (const char *arg)       /* bytes per sample */
      mus_long_t mus_sound_data_location (const char *arg)  /* location of first sample (bytes) */
      int mus_sound_bits_per_sample(const char *arg)   /* bits per sample */
      int mus_bytes_per_sample(int format)             /* bytes per sample */

      int mus_sound_chans (const char *arg)            /* number of channels (samples are interleaved) */
      int mus_sound_srate (const char *arg)            /* sampling rate */

      int mus_sound_header_type (const char *arg)      /* header type (aiff etc) */
      int mus_sound_data_format (const char *arg)      /* data format (alaw etc) */
      int mus_sound_original_format (const char *arg)  /* unmodified data format specifier */
      int mus_sound_type_specifier (const char *arg)   /* original header type identifier */

      char *mus_sound_comment (const char *arg)        /* comment if any */
      mus_long_t mus_sound_comment_start (const char *arg)  /* comment start (bytes) if any */
      mus_long_t mus_sound_comment_end (const char *arg)    /* comment end (bytes) */
      int *mus_sound_loop_info(const char *arg)        /* 8 loop vals (mode,start,end) then base-detune and base-note  (empty list if no loop info found) */

      int mus_sound_write_date (const char *arg)       /* bare (uninterpreted) file write date */

      int mus_sound_initialize(void)                   /* initialize everything */

The following can be used to provide user-understandable descriptions of the header type and the data format:

      char *mus_header_type_name(int type)             /* "AIFF" etc */
      char *mus_data_format_name(int format)           /* "16-bit big endian linear" etc */
      char *mus_header_type_to_string(int type)
      char *mus_data_format_to_string(int format)
      const char *mus_data_format_short_name(int format)

In all cases if an error occurs, -1 (MUS_ERROR) is returned, and some sort of error message is printed; to customize error handling, use mus_set_error_handler and mus_set_print_handler.

      mus_error_handler_t *mus_error_set_handler(mus_error_handler_t *new_error_handler);
      mus_print_handler_t *mus_print_set_handler(mus_print_handler_t *new_print_handler);

To decode the error indication, use:

      char *mus_error_to_string(int err);

Header data is cached internally, so the actual header is read only if it hasn't already been read, or the write date has changed. Loop points are also available, if there's interest. To go below the "sound" level, see headers.c — once a header has been read, all the components that have been found can be read via functions such as mus_header_srate.

Data

The following functions provide access to sound file data:

      int mus_sound_open_input (const char *arg)
      int mus_sound_open_output (const char *arg, int srate, int chans, int data_format, int header_type, const char *comment)
      int mus_sound_reopen_output (const char *arg, int type, int format, mus_long_t data_loc)
      int mus_sound_close_input (int fd)
      int mus_sound_close_output (int fd, mus_long_t bytes_of_data)
      int mus_sound_read (int fd, int beg, int end, int chans, mus_sample_t **bufs)
      int mus_sound_write (int fd, int beg, int end, int chans, mus_sample_t **bufs)
      mus_long_t mus_sound_seek_frame (int fd, mus_long_t frame)

mus_sample_t defaults to float, but can also be int. It is set when sndlib is built, and refers to Sndlib's internal representation of sample values. There are corresponding macros to convert from the sample type to C types (MUS_SAMPLE_TO_FLOAT, etc), and the reverse (MUS_FLOAT_TO_SAMPLE, etc).

mus_sound_open_input opens arg for reading. Most standard uncompressed formats are readable. This function returns the associated file number, or -1 upon failure.

mus_sound_close_input closes an open sound file. Its argument is the integer returned by mus_sound_open_input.

mus_sound_open_output opens (creates) the file arg, setting its sampling rate to be srate, number of channels to chans, data format to data_format (see sndlib.h for these types: MUS_BSHORT, means 16-bit 2's complement big endian fractions), header type to header_type (AIFF for example; the available writable header types are MUS_AIFC (or AIFF), MUS_RIFF ('wave'), MUS_RF64, MUS_NEXT, MUS_NIST, MUS_CAFF, and MUS_IRCAM), and comment (if any) to comment. The header is not considered complete without an indication of the data size, but since this is rarely known in advance, it is supplied when the sound file is closed. mus_sound_open_output function returns the associated file number.

mus_sound_close_output first updates the file's header to reflect the final data size bytes_of_data, then closes the file. The argument fd is the integer returned by mus_sound_open_output.

mus_sound_read reads data from the file indicated by fd, placing data in the array obufs as mus_sample_t values (floats normally). chans determines how many arrays of samples are in obufs, which is filled by mus_sound_read from its index beg to end with zero padding if necessary.

mus_sound_write writes samples to the file indicated by fd, starting for each of chans channels in obufs at beg and ending at end.

mus_sound_seek_frame moves the read or write position for the file indicated by fd to the desired frame.

Hardware

The following functions provide access to audio harware. If an error occurs, they return -1 (MUS_ERROR).

      int mus_audio_initialize(void)
      int mus_audio_open_output(int dev, int srate, int chans, int format, int size)
      int mus_audio_open_input(int dev, int srate, int chans, int format, int size)
      int mus_audio_write(int line, char *buf, int bytes)
      int mus_audio_close(int line)
      int mus_audio_read(int line, char *buf, int bytes)

mus_audio_initialize takes care of any necessary initialization.

mus_audio_open_input opens an audio port to read sound data (i.e. a microphone, line in, etc). The input device is dev (see sndlib.h for details; when in doubt, use MUS_AUDIO_DEFAULT). The input sampling rate is srate or as close as we can get to it. The number of input channels (if available) is chans. The input data format is format (when in doubt, use the macro MUS_AUDIO_COMPATIBLE_FORMAT). And the input buffer size (if settable at all) is size (bytes). This function returns an integer to distinguish its port from others that might be in use.

mus_audio_open_output opens an audio port to write data (i.e. speakers, line out, etc). The output device is dev (see sndlib.h). Its sampling rate is srate, number of channels chans, data format format, and buffer size size. This function returns the associated line number of the output port.

mus_audio_close closes the port (input or output) associated with line.

mus_audio_read reads sound data from line. The incoming 'bytes' bytes of data are placed in buf. If no error was returned from mus_audio_open_input, the data is in the format requested by that function with channels interleaved.

mus_audio_write writes 'bytes' bytes of data in buf to the output port associated with line. This data is assumed to be in the format requested by mus_audio_open_output with channels interleaved.

Music V

clm.c and friends implement all the generators found in CLM, a music V implementation, and clm2xen.c ties these into the languages supported by the xen package (currently s7, Ruby, and Forth). The primary clm documentation (which describes both the Scheme and Common Lisp implementations) is clm.html found in clm-4.tar.gz or sndclm.html in snd-12.tar.gz alongside sndlib at ccrma-ftp. The simplest way to try these out is to load them into Snd; see extsnd.html, examp.scm, and snd-test.scm in snd-12.tar.gz for more details. The following briefly describes the C calls (see clm.h).

clm.c implements a bunch of generators and sound IO handlers. Each generator has three associated functions, make-gen, gen, and gen_p; the first creates the generator (if needed), the second gets the next sample from the generator, and the last examines some pointer to determine if it is that kind of generator. In addition, there are a variety of generic functions that generators respond to: mus_free, for example, frees a generator, and mus_frequency returns its current frequency, if relevant. All generators are pointers to mus_any structs. Finally, CLM has two special data types: frame and mixer. A frame is an array that represents a multichannel sample (that is, in a stereo file, at time 0.0, there are two samples, one for each channel). A mixer is a array of arrays that represents a set of input and output scalers, as if it were the current state of a mixing console's volume controls. A frame (a multichannel input) can be mixed into a new frame (a multichannel output) by passing it through a mixer (a matrix, the operation being a matrix multiply).

- oscil — generate a sine wave.
  - mus_any *mus_make_oscil (float freq, float phase)
  - float mus_oscil (mus_any *o, float fm, float pm)
  - int mus_oscil_p (mus_any *ptr)

        mus_any *osc;
        osc = mus_make_oscil(440.0, 0.0);
        if (mus_oscil_p(osc))
          fprintf(stderr, "%.3f, %.3f ", .1 * mus_oscil(osc, 0.0, 0.0), mus_frequency(osc));
        mus_free(osc);

The other generators are:

- sum_of_cosines: generate a pulse train made up of cosines
- sum_of_sines: generate a sum of sines
- delay: a delay line with optional interpolation
- tap: read delay line
- comb: comb filter
- notch: notch filter
- all_pass: all pass filter
- table_lookup: interpolating table lookup
- sawtooth_wave, triangle_wave, pulse_train, square_wave
- rand: white noise (a step function)
- rand-interp: interpolating noise
- asymmetric_fm: a variety of FM
- one_zero, two_zero, one_pole, two_pole: basic filters
- formant: create a formant region (two poles, two zeros)
- sine_summation: another way to create sine waves
- filter, fir_filter, iir_filter: direct form filters of any order
- wave_train: sequence of possibly overlapping waves
- env: envelopes
- polyshape, polywave: waveshaping
- readin, file_to_sample, file_to_frame, in_any: file sample input
- locsig, sample_to_file, frame_to_file, out_any: file sample output
- src: sampling rate conversion
- granulate: granular synthesis
- convolve: convolution
- phase-vocoder: phase vocoder
- moving-average: moving window average
- ssb-am: single side-bank amplitude modulation

Some useful functions provided by clm.c are:

- float mus_radians_to_hz(float rads): convert radians/sample to cycles/sec.
- float mus_hz_to_radians(float hz): and the reverse.
- float mus_degrees_to_radians(float deg): convert degrees to radians.
- float mus_radians_to_degrees(float rads): and the reverse.
- float mus_srate(void): current sampling rate
- float mus_set_srate(float rate): set current sampling rate
- float mus_ring_modulate(float sig1, float sig2): multiply sig1 by sig2
- float mus_amplitude_modulate(float s1, float s2, float s3): AM
- float mus_contrast_enhancement(float sig, float index)
- float mus_dot_product(float *data1, float *data2, int size)
- void mus_clear_array(float *arr, int size)
- float mus_array_interp(float *wave, float phase, int size)
- float mus_polynomial(float *coeffs, float x, int ncoeffs);
- void mus_multiply_arrays(float *data, float *window, int len);
- void mus_rectangular_to_polar(float *rl, float *im, int size);
- void mus_spectrum(float *rdat, float *idat, float *window, int n, int type)
- void mus_fft(float *rl, float *im, int n, int isign)
- float *mus_make_fft_window(int size, int type, float beta)
- void mus_convolution(float* rl1, float* rl2, int n, int ipow)
- float *mus_partials_to_wave(float *partial_data, int partials, float *table, int table_size, int normalize)
- float *mus_phase_partials_to_wave(float *partial_data, int partials, float *table, int table_size, int normalize)
- float mus_samples_to_seconds(mus_long_t samps)
- mus_long_t mus_seconds_to_samples(float secs)

and various others: see clm.h.

The more useful generic functions are:

- int mus_free(mus_any *ptr)
- char *mus_describe(mus_any *gen)
- float mus_phase(mus_any *gen)
- float mus_set_phase(mus_any *gen, float val)
- float mus_set_frequency(mus_any *gen, float val)
- float mus_frequency(mus_any *gen)
- float mus_run(mus_any *gen, float arg1, float arg2)
- int mus_length(mus_any *gen)
- int mus_set_length(mus_any *gen, int len)
- float *mus_data(mus_any *gen)
- float *mus_set_data(mus_any *gen, float *data)
- char *mus_name(mus_any *ptr)
- float mus_scaler(mus_any *gen)
- float mus_set_scaler(mus_any *gen, float val)
- float mus_apply(mus_any *gen, ...)

Errors are reported through mus_error which can be redirected or muffled. See clm2xen.c for an example.

Examples
sndinfo

This program prints out a description of a sound file (sndinfo.c).

    int main(int argc, char *argv[])
    {
      int fd, chans, srate;
      mus_long_t samples;
      float length;
      time_t date;
      char *comment;
      char timestr[64];
      mus_sound_initialize();	    /* initialize sndlib */
      fd = mus_file_open_read(argv[1]); /* see if it exists */
      if (fd != -1)
        {
          close(fd);
          date = mus_sound_write_date(argv[1]);
          srate = mus_sound_srate(argv[1]);
          chans = mus_sound_chans(argv[1]);
          samples = mus_sound_samples(argv[1]);
          comment = mus_sound_comment(argv[1]);
          length = (double)samples / (float)(chans * srate);
          strftime(timestr, 64, "%a %d-%b-%y %H:%M %Z", localtime(&date));
          fprintf(stdout, "%s:\n  srate: %d\n  chans: %d\n  length: %f\n",
    	      argv[1], srate, chans, length);
          fprintf(stdout, "  type: %s\n  format: %s\n  written: %s\n  comment: %s\n",
    	      mus_header_type_name(mus_sound_header_type(argv[1])),
    	      mus_data_format_name(mus_sound_data_format(argv[1])),
    	      timestr, comment);
        }
      else
        fprintf(stderr, "%s: %s\n", argv[1], strerror(errno));
      return(0);
    }

sndplay

This code plays a sound file (sndplay.c):

    int main(int argc, char *argv[])
    {
      int fd, afd, i, j, n, k, chans, srate, outbytes;
      mus_long_t frames;
      mus_sample_t **bufs;
      short *obuf;
      mus_sound_initialize();
      fd = mus_sound_open_input(argv[1]);
      if (fd != -1)
        {
          chans = mus_sound_chans(argv[1]);
          srate = mus_sound_srate(argv[1]);
          frames = mus_sound_frames(argv[1]);
          outbytes = BUFFER_SIZE * chans * 2;
          bufs = (mus_sample_t **)calloc(chans, sizeof(mus_sample_t *));
          for (i=0;i<chans;i++)
            bufs[i] = (mus_sample_t *)calloc(BUFFER_SIZE, sizeof(mus_sample_t));
          obuf = (short *)calloc(BUFFER_SIZE * chans, sizeof(short));
          afd = mus_audio_open_output(MUS_AUDIO_DEFAULT, srate, chans, MUS_AUDIO_COMPATIBLE_FORMAT, outbytes);
          if (afd != -1)
    	{
    	  for (i = 0; i < frames; i += BUFFER_SIZE)
    	    {
    	      mus_sound_read(fd, 0, BUFFER_SIZE - 1, chans, bufs);
    	      for (k = 0, j = 0; k < BUFFER_SIZE; k++, j += chans)
    		for (n = 0; n < chans; n++)
                      obuf[j + n] = MUS_SAMPLE_TO_SHORT(bufs[n][k]);
    	      mus_audio_write(afd, (char *)obuf, outbytes);
    	    }
    	  mus_audio_close(afd);
    	}
          mus_sound_close_input(fd);
          for (i = 0; i < chans; i++) free(bufs[i]);
          free(bufs);
          free(obuf);
        }
      return(0);
    }

sndsine

This program writes a one channel NeXT/Sun sound file containing a sine wave at 440 Hz.

    int main(int argc, char *argv[])
    {
      int fd, i, k, frames;
      float phase, incr;
      mus_sample_t *obuf[1];
      mus_sound_initialize();
      fd = mus_sound_open_output(argv[1], 22050, 1, MUS_BSHORT, MUS_NEXT, "created by sndsine");
      if (fd != -1)
        {
          frames = 22050;
          phase = 0.0;
          incr = 2 * M_PI * 440.0 / 22050.0;
          obuf[0] = (mus_sample_t *)calloc(BUFFER_SIZE, sizeof(mus_sample_t));
          k = 0;
          for (i = 0; i < frames; i++)
    	{
    	  obuf[0][k] = MUS_FLOAT_TO_SAMPLE(0.1 * sin(phase)); /* amp = .1 */
    	  phase += incr;
    	  k++;
    	  if (k == BUFFER_SIZE)
    	    {
    	      mus_sound_write(fd, 0, BUFFER_SIZE-1, 1, obuf);
    	      k=0;
    	    }
    	}
          if (k > 0) mus_sound_write(fd, 0, k - 1, 1, obuf);
          mus_sound_close_output(fd, 22050 * mus_bytes_per_sample(MUS_BSHORT));
          free(obuf[0]);
        }
      return(0);
    }

clmosc

This is program uses the clm.c oscillator and output functions to write the same sine wave as we wrote in SndSine.

    int main(int argc, char *argv[])
    {
      int i;
      mus_any *osc, *op;
      mus_sound_initialize();
      osc = mus_make_oscil(440.0, 0.0);
      op = mus_make_sample_to_file("test.snd", 1, MUS_BSHORT, MUS_NEXT);
      if (op)
        for (i = 0; i < 22050; i++)
          mus_sample_to_file(op, i, 0, .1 * mus_oscil(osc, 0.0, 0.0));
      mus_free(osc);
      if (op) mus_free(op);
      return(0);
    }

Here is the fm-violin and a sample with-sound call:

    static int feq(float x, int i) {return(fabs(x-i)<.00001);}

    void fm_violin(float start, float dur, float frequency, float amplitude, float fm_index, mus_any *op)
    {
     float pervibfrq = 5.0,
       ranvibfrq = 16.0,
       pervibamp = .0025,
       ranvibamp = .005,
       noise_amount = 0.0,
       noise_frq = 1000.0,
       gliss_amp = 0.0,
       fm1_rat = 1.0,
       fm2_rat = 3.0,
       fm3_rat = 4.0,
       reverb_amount = 0.0,
       degree = 0.0,
       distance = 1.0;
      float fm_env[] = {0.0, 1.0, 25.0, 0.4, 75.0, 0.6, 100.0, 0.0};
      float amp_env[] = {0.0, 0.0,  25.0, 1.0, 75.0, 1.0, 100.0, 0.0};
      float frq_env[] = {0.0, -1.0, 15.0, 1.0, 25.0, 0.0, 100.0, 0.0};
      int beg = 0, end, easy_case = 0, npartials, i;
      float *coeffs, *partials;
      float frq_scl, maxdev, logfrq, sqrtfrq, index1, index2, index3, norm;
      float vib = 0.0, modulation = 0.0, fuzz = 0.0, indfuzz = 1.0;
      mus_any *carrier, *fmosc1, *fmosc2, *fmosc3, *ampf;
      mus_any *indf1, *indf2, *indf3, *fmnoi = NULL, *pervib, *ranvib, *frqf = NULL, *loc;
      beg = start * mus_srate();
      end = beg + dur * mus_srate();
      frq_scl = mus_hz_to_radians(frequency);
      maxdev = frq_scl * fm_index;
      if ((noise_amount == 0.0) &&
          (feq(fm1_rat, floor(fm1_rat))) &&
          (feq(fm2_rat, floor(fm2_rat))) &&
          (feq(fm3_rat, floor(fm3_rat))))
        easy_case = 1;
      logfrq = log(frequency);
      sqrtfrq = sqrt(frequency);
      index1 = maxdev * 5.0 / logfrq;
      if (index1 > M_PI) index1 = M_PI;
      index2 = maxdev * 3.0 * (8.5 - logfrq) / (3.0 + frequency * .001);
      if (index2 > M_PI) index2 = M_PI;
      index3 = maxdev * 4.0 / sqrtfrq;
      if (index3 > M_PI) index3 = M_PI;
      if (easy_case)
        {
          npartials = floor(fm1_rat);
          if ((floor(fm2_rat)) > npartials) npartials = floor(fm2_rat);
          if ((floor(fm3_rat)) > npartials) npartials = floor(fm3_rat);
          npartials++;
          partials = (float *)calloc(npartials, sizeof(float));
          partials[(int)(fm1_rat)] = index1;
          partials[(int)(fm2_rat)] = index2;
          partials[(int)(fm3_rat)] = index3;
          coeffs = mus_partials_to_polynomial(npartials, partials, 1);
          norm = 1.0;
        }
      else norm = index1;
      carrier = mus_make_oscil(frequency, 0.0);
      if (easy_case == 0)
        {
          fmosc1 = mus_make_oscil(frequency * fm1_rat, 0.0);
          fmosc2 = mus_make_oscil(frequency * fm2_rat, 0.0);
          fmosc3 = mus_make_oscil(frequency * fm3_rat, 0.0);
        }
      else fmosc1 = mus_make_oscil(frequency, 0.0);
      ampf = mus_make_env(amp_env, 4, amplitude, 0.0, 1.0, dur, 0, NULL);
      indf1 = mus_make_env(fm_env, 4, norm, 0.0, 1.0, dur, 0, NULL);
      if (gliss_amp != 0.0)
        frqf = mus_make_env(frq_env, 4, gliss_amp * frq_scl, 0.0, 1.0, dur, 0, NULL);
      if (easy_case == 0)
        {
          indf2 = mus_make_env(fm_env, 4, index2, 0.0, 1.0, dur, 0, NULL);
          indf3 = mus_make_env(fm_env, 4, index3, 0.0, 1.0, dur, 0, NULL);
        }
      pervib = mus_make_triangle_wave(pervibfrq, frq_scl * pervibamp, 0.0);
      ranvib = mus_make_rand_interp(ranvibfrq, frq_scl * ranvibamp);
      if (noise_amount != 0.0) fmnoi = mus_make_rand(noise_frq, noise_amount * M_PI);
      loc = mus_make_locsig(degree, distance, reverb_amount, 1, (mus_any *)op, 0, NULL, MUS_INTERP_LINEAR);
      for (i = beg; i < end; i++)
        {
          if (noise_amount != 0.0) fuzz = mus_rand(fmnoi, 0.0);
          if (frqf) vib = mus_env(frqf); else vib = 0.0;
          vib += mus_triangle_wave(pervib, 0.0) + mus_rand_interp(ranvib, 0.0);
          if (easy_case)
    	modulation = mus_env(indf1) *
                         mus_polynomial(coeffs, mus_oscil(fmosc1, vib, 0.0), npartials);
          else
    	modulation = mus_env(indf1) * mus_oscil(fmosc1, (fuzz + fm1_rat * vib), 0.0) +
    	             mus_env(indf2) * mus_oscil(fmosc2, (fuzz + fm2_rat * vib), 0.0) +
    	             mus_env(indf3) * mus_oscil(fmosc3, (fuzz + fm3_rat * vib), 0.0);
          mus_locsig(loc, i, mus_env(ampf) * mus_oscil(carrier, vib + indfuzz * modulation, 0.0));
        }
      mus_free(pervib);
      mus_free(ranvib);
      mus_free(carrier);
      mus_free(fmosc1);
      mus_free(ampf);
      mus_free(indf1);
      if (fmnoi) mus_free(fmnoi);
      if (frqf) mus_free(frqf);
      if (!(easy_case))
        {
          mus_free(indf2);
          mus_free(indf3);
          mus_free(fmosc2);
          mus_free(fmosc3);
        }
      else
        free(partials);
      mus_free(loc);
    }

    int main(int argc, char *argv[])
    {
      mus_any *op = NULL;
      mus_sound_initialize();
      op = mus_make_sample_to_file("test.snd", 1, MUS_BSHORT, MUS_NEXT);
      if (op)
        {
          fm_violin(0.0, 20.0, 440.0, .3, 1.0, op);
          mus_free(op);
        }
      return(0);
    }

The CLM version is v.ins, the Scheme version can be found in v.scm, and the Ruby version is in v.rb. This code can be run:

    cc v.c -o vc -O3 -lm io.o headers.o audio.o sound.o clm.o -DLINUX

For generators such as src that take a function for "as-needed" input, you can use something like:

    static mus_float_t input_as_needed(void *arg, int dir) {/* get input here — arg is "sf" passed below */}

    static SCM call_phase-vocoder(void)
    {
      mus_any *pv;
      int sf; /* file channel or whatever */
      pv = mus_make_phase_vocoder(NULL, 512, 4, 128, 0.5, NULL, NULL, NULL, (void *)sf);
      mus_phase_vocoder(pv, &input_as_needed);
      /* etc */
    }

Michael Scholz has written a package using these functions, and several CLM instruments: see the sndins directory, and in particular the README file, for details.

Other examples

The primary impetus for the sound library was the development of Snd and CLM, both of which are freely available.

Extension Languages

Much of sndlib is accessible at run time in any program that has one of the languages supported by the xen package (s7, Ruby, Forth); the modules sndlib2xen and clm2xen tie most of the library into that language making it possible to call the library functions from its interpreter. The documentation is scattered around, unfortunately: the clm side is in sndclm.html and extsnd.html with many examples in Snd's examp.scm. Most of these are obvious translations of the constants and functions described above into Scheme. To initialize sndlib, call Init_sndlib.

    (let ((sndlib (dynamic-link "libsndlib.so")))
      (dynamic-call (dynamic-func "Init_sndlib" sndlib) #f))

Init_sndlib ties most of the functions mentioned above into the extension language (s7, Forth, or Ruby).

      mus-next mus-aifc mus-rf64 mus-riff mus-nist mus-raw mus-ircam mus-aiff mus-bicsf mus-soundfont mus-voc mus-svx mus-caff

      mus-bshort mus-lshort mus-mulaw mus-alaw mus-byte mus-ubyte mus-bfloat
      mus-lfloat mus-bint mus-lint mus-b24int mus-l24int mus-bdouble mus-ldouble
      mus-ubshort mus-ulshort

      mus-sound-samples (filename)             samples of sound according to header (can be incorrect)
      mus-sound-frames (filename)              frames of sound according to header (can be incorrect)
      mus-sound-duration (filename)            duration of sound in seconds
      mus-sound-datum-size (filename)          bytes per sample
      mus-sound-data-location (filename)       location of first sample (bytes)
      mus-sound-chans (filename)               number of channels (samples are interleaved)
      mus-sound-srate (filename)               sampling rate
      mus-sound-header-type (filename)         header type (e.g. mus-aiff)
      mus-sound-data-format (filename)         data format (e.g. mus-bshort)
      mus-sound-length (filename)              true file length (bytes)
      mus-sound-type-specifier (filename)      original header type identifier
      mus-sound-maxamp(filename)               returns a list of max amps and locations thereof
      mus-sound-loop-info(filename)            returns list of 4 loop values (the actual mark positions here, not
                                               the so-called id's), then base-note and base-detune

      mus-header-type-name (type)              e.g. "AIFF"
      mus-data-format-name (format)            e.g. "16-bit big endian linear"
      mus-sound-comment (filename)             header comment, if any
      mus-sound-write-date (filename)          sound write date
      data-format-bytes-per-sample (format)    bytes per sample

      mus-audio-reinitialize                   force re-check of available audio devices

      mus-sound-open-input (filename)          open filename (a sound file) returning an integer ("fd" below)
      mus-sound-open-output (filename srate chans data-format header-type comment)
                                               create a new sound file with the indicated attributes, return "fd"
      mus-sound-reopen-output (filename chans data-format header-type data-location)
                                               reopen (without disturbing) filename, ready to be written
      mus-sound-close-input (fd)               close sound file
      mus-sound-close-output (fd bytes)        close sound file and update its length indication, if any
      mus-sound-read (fd beg end chans sdata)  read data from sound file fd loading the data array from beg to end
                                               sdata is a sound-data object that should be able to accommodate the read
      mus-sound-write (fd beg end chans sdata) write data to sound file fd
      mus-sound-seek-frame (fd frame)          move to frame in sound file fd
      mus-file-clipping (fd)                   whether output is clipped in file 'fd'
      mus-clipping ()                          global clipping choice

      mus-audio-open-output (device srate chans format bufsize)
                                               open audio port device ready for output with the indicated attributes
      mus-audio-open-input (device srate chans format bufsize)
                                               open audio port device ready for input with the indicated attributes
      mus-audio-write (line sdata frames)      write frames of data from sound-data object sdata to port line
      mus-audio-read (line sdata frames)       read frames of data into sound-data object sdata from port line
      mus-audio-close (line)                   close audio port line

      mus-oss-set-buffers (num size)           in Linux (OSS) sets the number and size of the OSS "fragments"

      make-sound-data (chans, frames)      return a sound-data object with chans arrays, each of length frames
      sound-data-ref (obj chan frame)      return (as a float) the sample in channel chan at location frame
      sound-data-set! (obj chan frame val) set obj's sample at frame in chan to (the float) val
      sound-data? (obj)                    #t if obj is of type sound-data
      sound-data-length (obj)              length of each channel of data in obj
      sound-data-chans (obj)               number of channels of data in obj
      sound-data->vct (sdobj chan vobj)    place sound-data channel data in vct
      vct->sound-data (vobj sdobj chan)    place vct data in sound-data

    ;;; this function prints header information
    (define info
      (lambda (file)
        (string-append
         file
         ": chans: " (number->string (mus-sound-chans file))
         ", srate: " (number->string (mus-sound-srate file))
         ", " (mus-header-type-name (mus-sound-header-type file))
         ", " (mus-data-format-name (mus-sound-data-format file))
         ", len: " (number->string
                    (/ (mus-sound-samples file)
                       (* (mus-sound-chans file) (mus-sound-srate file)))))))

    ;;; this function reads the first 32 samples of a file, returning the 30th in channel 0
    (define read-sample-30
      (lambda (file)
        (let* ((fd (mus-sound-open-input file))
    	   (chans (mus-sound-chans file))
    	   (data (make-sound-data chans 32)))
          (mus-sound-read fd 0 31 chans data)
          ;; we could use sound-data->vct here to return all the samples
          (let ((val (sound-data-ref data 0 29)))
    	(mus-sound-close-input fd)
    	val))))

related documentation: snd.html grfsnd.html extsnd.html sndscm.html sndclm.html libxm.html s7.html index.html