SndLib

                 Bill Schottstaedt (bil@ccrma.stanford.edu)
                  snd.html extsnd.html grfsnd.html clm.html

Contents

   * Introduction
   * Headers
   * Data
   * Hardware
   * Music V
   * Examples
        o SndInfo
        o SndPlay
        o SndRecord
        o AudInfo
        o SndSine
        o clmosc
        o Other Examples
   * How to Make Sndlib and the examples
   * Current Status
   * Lower Levels
   * Sndlib and Guile

Introduction

The sound library is a collection of sound file and audio hardware handlers
written in C and running currently on SGI (either audio library), NeXT, Sun,
Be, OSS or ALSA (Linux and others), Mac, HPUX, LinuxPPC, and Windoze
systems. It provides relatively straightforward access to many sound file
headers and data types, and most of the features of the audio hardware.

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)
   * sndlib2scm.c and sndlib-strings.h (tie preceding into Guile)
   * clm.c and clm.h (Music V implementation)
   * clm2scm.c, vct.c and vct.h (tie clm.c into Guile)
   * old-sndlib.h (old names)

In version 6 and then to a greater extent in version 10, I changed the
exported names to use the prefix "mus" or "MUS"; see old-sndlib.h for
backwards compatibility; also, if you have common lisp, you can
automatically translate files from version 9 or earlier to version 10 by
loading transnd.cl and calling the translate function. The naming scheme is
more as less as follows: constants start with "MUS_", audio hardware
constants 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".

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

  ./configure
  make

To install it, 'make install' -- I've tested this process in Linux, SGI,
Sun, and NeXT. It could conceivably work elsewhere. For more details see How
to Make Sndlib below.

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.

  int mus_sound_samples (const char *arg)          /* samples of sound according to header (can be incorrect) */
  int mus_sound_frames (const char *arg)           /* samples per channel */
  float mus_sound_duration (const char *arg)
  int mus_sound_datum_size (const char *arg)       /* bytes per sample */
  int mus_sound_data_location (const char *arg)    /* location of first sample (bytes) */
  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 */
  char *mus_sound_comment (const char *arg)        /* comment if any */
  int mus_sound_comment_start (const char *arg)    /* comment start (bytes) if any */
  int mus_sound_comment_end (const char *arg)      /* comment end (bytes) */
  int mus_sound_length (const char *arg)           /* true file length (for error checks) */
  int mus_sound_fact_samples (const char *arg)     /* compression scheme data */
  int mus_sound_distributed (const char *arg)      /* is header scattered around in sound file */
  int mus_sound_write_date (const char *arg)       /* bare (uninterpreted) file write date */
  int mus_sound_type_specifier (const char *arg)   /* original header type identifier */
  int mus_sound_align (const char *arg)            /* more compression data */
  int mus_sound_bits_per_sample(const char *arg)   /* bits per sample */
  int mus_data_format_to_bytes_per_sample(int format) /* bytes per sample */
  int mus_sound_max_amp(const char *arg, int *vals)/* return list of max-amp sample pairs */
  int mus_sound_aiff_p(const char *arg)            /* is it an old-style AIFF file (not AIFC) */
  int *mus_sound_loop_info(const char *arg)        /* 6 loop vals (mode,start,end) then base-detune and base-note  (empty list if no loop info found) */
  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 */

In all cases if an error occurs, -1 is returned; for information about the
error use:

  int mus_audio_error(void)                 /* returns error code indicated by preceding audio call */
  char *mus_audio_error_name(int err)       /* gives string decription of error code */

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, int data_loc)
  int mus_sound_close_input (int fd)
  int mus_sound_close_output (int fd, int bytes_of_data)
  int mus_sound_read (int fd, int beg, int end, int chans, MUS_SAMPLE_TYPE **bufs)
  int mus_sound_write (int fd, int beg, int end, int chans, MUS_SAMPLE_TYPE **bufs)
  int mus_sound_seek (int fd, long offset, int origin)
  int mus_sound_seek_frame (int fd, int frame)

MUS_SAMPLE_TYPE defaults to int, but can also be float -- 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 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, for example, 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_NEXT,
MUS_NIST, 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.
This 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 32-bit integers in the host's byte order. chans determines
how many arrays of ints are in obufs, which is filled by mus_sound_read from
its index beg to end with zero padding if necessary. See the sndplay example
below if this is not obvious.

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

mus_sound_seek moves the read or write position for the file indicated by fd
to offset given the origin indication (both treated as in lseek). The new
actual position attained is returned. In both cases (the returned value and
offset), the output datum size is considered to be 2, no matter what it
really is. That is, use byte positions as if you were always reading and
writing 16-bit data, and mus_sound_seek will compensate if its actually
32-bit floats or whatever. Since this is impossible to understand, there's
also mus_sound_seek_frame which moves to the indicated frame.

Hardware

The following functions provide access to audio harware. If an error occurs,
they return -1, and the mus_audio_error functions can be used to find out
what went wrong.

  int mus_audio_initialize(void)
  void mus_audio_save(void)
  void mus_audio_restore(void)
  void mus_audio_describe(void)
  char *mus_audio_report(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)
  int mus_audio_mixer_read(int dev, int field, int chan, float *val)
  int mus_audio_mixer_write(int dev, int field, int chan, float *val)
  int mus_audio_systems(void)
  char *mus_audio_system_name(int system)
  void mus_audio_set_dsp_devices(int cards, int *dsps, int *mixers)

mus_audio_initialize takes care of any necessary intialization.

mus_audio_save saves the current audio hardware state.

mus_audio_restore restores the audio hardware to the last saved state.

mus_audio_describe prints to stdout a description of the current state of
the audio hardware. mus_audio_report returns the same description as a
string.

mus_audio_systems returns the number of separate and complete audio systems
(soundcards essentially) that are available. mus_audio_system_name returns
some user-recognizable name for the given card.

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_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. In this and other related functions, the
device has an optional second portion that refers to the soundcard or system
for that device. MUS_AUDIO_PACK_SYSTEM(n) refers to the nth such card, so
(SNDLIB_DAC_DEVICE | MUS_AUDIO_PACK_SYSTEM(1)) is the 2nd card's dac (the
default is system 0, the first card).

mus_audio_open_output opens an audio port to write date (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.

mus_audio_mixer_read and mus_audio_mixer_write are complicated. They get and
set the audio hardware state. The audio hardware is treated as a set of
"systems" (sound cards) each of which has a set of "devices" (dacs, adcs,
etc), with various "fields" that can be read or set (gain, channels active,
etc). For example, a microphone is called the MUS_AUDIO_MICROPHONE, and its
hardware gain setting (if any) is called the MUS_AUDIO_AMP. All gains are
considered to be linear between 0.0 and 1.0, so to set the microphone's
first channel amplitude to .5 (that is, the gain of the signal before it
reaches the analog-to-digital converter),

  float vals[1];
  vals[0]=0.5;
  mus_audio_mixer_write(MUS_AUDIO_MICROPHONE,MUS_AUDIO_AMP,0,vals);

Similarly

  mus_audio_mixer_read(MUS_AUDIO_MICROPHONE,MUS_AUDIO_AMP,0,vals);
  amp=vals[0];

returns the current gain in the float array vals. mus_audio_mixer_read can
also return a description of the currently available audio hardware.

If a requested operation is not implemented, -1 is returned, and
SNDLIB_AUDIO_ERROR is set to MUS_AUDIO_CANT_READ or MUS_AUDIO_CANT_WRITE. If
an error occurs during the requested operation, -1 is returned, and
SNDLIB_AUDIO_ERROR is set to MUS_AUDIO_READ_ERROR or MUS_AUDIO_WRITE_ERROR.
If some operation cannot be performed on the current hardware, -1 is
returned and SNDLIB_AUDIO_ERROR tries to indicate what portion of the
requested operation is impossible (MUS_AUDIO_SRATE_NOT_AVAILABLE,
MUS_AUDIO_FORMAT_NOT_AVAILABLE, and so on).

Systems

Each separate sound card is called a system, accessible via the device
argument through the macro MUS_AUDIO_PACK_SYSTEM(n). The count starts at 0
which is the default. The function mus_audio_systems returns how many such
cards are available. (Currently it returns more than one only on Linux
systems with multiple sound cards).

Devices

Each audio system has a set of available devices. To find out what is
available on a given system

  #define LIST_MAX_SIZE 32;
  float device_list[LIST_MAX_SIZE];
  mus_audio_mixer_read(MUS_AUDIO_PACK_SYSTEM(0),MUS_AUDIO_PORT,LIST_MAX_SIZE,device_list);

The list of available devices is returned in the device_list array, with the
number of the devices as device_list[0]. The set of device identifiers is in
sndlib.h (MUS_AUDIO_LINE_IN for example). Two special devices are
MUS_AUDIO_MIXER and MUS_AUDIO_DAC_FILTER. The latter refers to the low-pass
filter often associated with a DAC. The former refers to a set of analog
gain and tone controls often associated with a sound card. The individual
gains are accessed through the various fields (described below).

Fields

The field argument in mus-audio-mixer-read and mus-audio-mixer-write selects
one aspect of the given card's devices' controls. The simplest operations
involve MUS_AUDIO_AMP and MUS_AUDIO_SRATE. The latter gets or sets the
sampling rate of the device, and the former gets or sets the amplitude
(between 0.0 and 1.0) of the specified channel of the device. The value to
be set or returned is in the 0th element of the vals array. An example of
reading the current microphone gain is given above. The meaning of the field
argument can depend on which device it is applied to, so there is some
complexity here. The channel argument usually selects which channel we are
interested in, but in some cases it instead tells mus-audio-mixer-read how
big a returned list can get. A brief description of the fields:

MUS_AUDIO_AMP       gain or volume control (0.0 to 1.0)
MUS_AUDIO_SRATE     sampling rate
MUS_AUDIO_CHANNEL   active channels

MUS_AUDIO_BASS, MUS_AUDIO_TREBLE    mixer's tone control
MUS_AUDIO_LINE                      mixer's line-in gain control
MUS_AUDIO_MICROPHONE                mixer's microphone gain control
similarly for MUS_AUDIO_IMIX, MUS_AUDIO_IGAIN,
              MUS_AUDIO_RECLEV, MUS_AUDIO_PCM, MUS_AUDIO_PCM2,
              MUS_AUDIO_OGAIN, MUS_AUDIO_LINE1,
              MUS_AUDIO_LINE2, MUS_AUDIO_LINE3, MUS_AUDIO_SYNTH

MUS_AUDIO_FORMAT    return list of usable sound formats (e.g. MUS_BSHORT)
MUS_AUDIO_PORT      return list of available devices (e.g. MUS_AUDIO_MICROPHONE)

MusicV

clm.c and friends implement all the generators found in CLM, a common lisp
music V implementation, and clm2scm.c ties these into Guile (Scheme). The
primary clm documentation (which describes both the Scheme and Common Lisp
implementations) is clm.html found in clm-2.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-3.tar.gz for more details.
The C implementation is essentially the same as the two Lisp versions, but
(as might be expected), works at a lower level, expecting the caller to
handle garbage collection and so forth. 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 multi-channel 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 multi-channel input) can be "mixed" into a new frame (a
multi-channel output) by passing it through a "mixer" (a matrix, the
operation being a matrix multiply).

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

       mus_any *osc;
       init_mus_module();
       osc = mus_make_oscil(440.0,0.0);
       if (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
   * 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
   * buffer -- a way to handle block processing in the generator world
   * env -- envelopes
   * waveshape -- waveshaping
   * readin, file2sample, file2frame, in_any -- file sample input
   * locsig, sample2file, frame2file, out_any -- file sample output
   * src -- sampling rate conversion
   * granulate -- granular synthesis
   * convolve -- convolution

Some useful functions provided by clm.c are:

   * float mus_radians2hz(float rads) -- convert radians/sample to
     cycles/sec.
   * float mus_hz2radians(float hz) -- and the reverse.
   * float mus_degrees2radians(float deg) -- convert degrees to radians.
   * float mus_radians2degrees(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_rectangular2polar(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, int ipow)
   * float *mus_make_fft_window(int size, int type, float beta)
   * void mus_convolution(float* rl1, float* rl2, int n, int ipow)
   * float *mus_partials2wave(float *partial_data, int partials, float
     *table, int table_size, int normalize)
   * float *mus_phasepartials2wave(float *partial_data, int partials, float
     *table, int table_size, int normalize)

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)
   * 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)
   * int mus_type(mus_any *ptr)
   * float mus_scaler(mus_any *gen)
   * float mus_set_scaler(mus_any *gen, float val)

Before using any of these functions, call init_mus_module. Errors are
reported through mus_error which can be redirected or muffled. See clm2scm.c
for an example.

  ------------------------------------------------------------------------

Examples

In the following examples I've omitted the usual garrulous C-header gab and
other inessential stuff. The full program code is available as noted below.

SndInfo

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

int main(int argc, char *argv[])
{
  int fd,chans,srate,samples;
  float length;
  time_t date;
  char *comment;
  char timestr[64];
  mus_sound_initialize();
  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 = (float)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,frames,outbytes;
  MUS_SAMPLE_TYPE **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_TYPE **)calloc(chans,sizeof(MUS_SAMPLE_TYPE *));
      for (i=0;i<chans;i++) bufs[i] = (MUS_SAMPLE_TYPE *)calloc(BUFFER_SIZE,sizeof(MUS_SAMPLE_TYPE));
      obuf = (short *)calloc(BUFFER_SIZE * chans,sizeof(short));
      afd = mus_audio_open_output(MUS_AUDIO_DEFAULT,srate,chans,MUS_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);
    }
  else
    fprintf(stderr,"%s: %s ",argv[1],mus_audio_error_name(mus_audio_error()));
  return(0);
}

SndRecord

This code records a couple seconds of sound from a microphone. Input formats
and sampling rates are dependent on available hardware, so in a "real"
program, you'd use mus_audio_mixer_read to find out what was available, then
float-sound to turn that data into a stream of floats. You'd also provide,
no doubt, some whizzy user interface to turn the thing off. (sndrecord.c)

int main(int argc, char *argv[])
{
  int fd,afd,i,err;
  short *ibuf;
#if MACOS
  argc = ccommand(&argv);
#endif
  afd = -1;
  mus_sound_initialize();
  fd = mus_sound_open_output(argv[1],22050,1,MUS_BSHORT,MUS_NEXT,"created by sndrecord");
  if (fd != -1)
    {
      ibuf = (short *)calloc(BUFFER_SIZE,sizeof(short));
      afd = mus_audio_open_input(MUS_AUDIO_MICROPHONE,22050,1,MUS_BSHORT,BUFFER_SIZE);
      if (afd != -1)
        {
          for (i=0;i<10;i++) /* grab 10 buffers of input */
            {
              err = mus_audio_read(afd,(char *)ibuf,BUFFER_SIZE*2);
              if (err != MUS_AUDIO_NO_ERROR) {fprintf(stderr,mus_audio_error_name(mus_audio_error())); break;}
              write(fd,ibuf,BUFFER_SIZE*2);
            }
          mus_audio_close(afd);
        }
      else
        fprintf(stderr,mus_audio_error_name(mus_audio_error()));
      mus_sound_close_output(fd,BUFFER_SIZE*10*2);
      free(ibuf);
    }
  else
    fprintf(stderr,"%s: %s ",argv[1],strerror(errno));
  return(0);
}

AudInfo

This program describes the current audio harware state (audinfo.c):

int main(int argc, char *argv[])
{
  mus_sound_initialize();
  mus_audio_describe();
  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_TYPE *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*PI*440.0/22050.0;
      obuf[0] = (MUS_SAMPLE_TYPE *)calloc(BUFFER_SIZE,sizeof(MUS_SAMPLE_TYPE));
      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_data_format_to_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();
  init_mus_module();
  osc = mus_make_oscil(440.0,0.0);
  op = mus_make_sample2file("test.snd",1,MUS_BSHORT,MUS_NEXT);
  if (op) for (i=0;i<22050;i++) mus_sample2file(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,vib = 0.0,modulation = 0.0,fuzz = 0.0,indfuzz = 1.0,ampfuzz = 1.0;
  mus_any *carrier,*fmosc1,*fmosc2,*fmosc3,*ampf,*indf1,*indf2,*indf3,*fmnoi = NULL,*pervib,*ranvib,*frqf = NULL,*loc;
  beg = start * mus_srate();
  end = beg + dur * mus_srate();
  frq_scl = mus_hz2radians(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_partials2polynomial(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,0,0,NULL);
  indf1 = mus_make_env(fm_env,4,norm,0.0,1.0,dur,0,0,NULL);
  if (gliss_amp != 0.0) frqf = mus_make_env(frq_env,4,gliss_amp * frq_scl,0.0,1.0,dur,0,0,NULL);
  if (easy_case == 0)
    {
      indf2 = mus_make_env(fm_env,4,index2,0.0,1.0,dur,0,0,NULL);
      indf3 = mus_make_env(fm_env,4,index3,0.0,1.0,dur,0,0,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_output *)op,NULL);
  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 == 0)
    {
      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 *osc = NULL,*op = NULL;
  mus_sound_initialize();
  init_mus_module();
  op = mus_make_sample2file("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 examp.scm. This
code can be run:

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

  ------------------------------------------------------------------------

Other Examples

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

  ------------------------------------------------------------------------

How to Make Sndlib and the examples

The Sndlib files can be used as separate modules or made into a library. The
following sequence, for example, builds the sndplay program from scratch on
an SGI:

cc -c io.c -O -DSGI
cc -c headers.c -O -DSGI
cc -c audio.c -O -DSGI
cc -c sound.c -O -DSGI
cc sndplay.c -o sndplay -O -DSGI audio.o io.o headers.o sound.o -laudio -lm

To make a library out of the sndlib files, first compile them as above,
then:

ld -r audio.o io.o headers.o sound.o -o sndlib.a
cc sndplay.c -o sndplay -O -DSGI sndlib.a -laudio -lm

The full sequence in Linux:

cc -c io.c -O -DLINUX
cc -c audio.c -O -DLINUX
cc -c headers.c -O -DLINUX
cc -c sound.c -O -DLINUX
cc sndplay.c -o sndplay -O -DLINUX audio.o io.o headers.o sound.o -lm

ld -r audio.o io.o headers.o sound.o -o sndlib.a
cc sndplay.c -o sndplay -O -DLINUX sndlib.a -lm

And on a NeXT:

cc -c io.c -O -DNEXT
cc -c audio.c -O -DNEXT
cc -c headers.c -O -DNEXT
cc -c sound.c -O -DNEXT
cc sndplay.c -o sndplay -O -DNEXT audio.o io.o headers.o sound.o

ld -r audio.o io.o headers.o sound.o -o sndlib.a
cc sndplay.c -o sndplay -O -DNEXT sndlib.a

Some similar sequence should work on a Sun (-DSOLARIS) or in HP-UX (-DHPUX).
On a Mac, you need to make a project in CodeWarrior or whatever that
includes all the basic sndlib .c and .h files (io.c, audio.c headers.c,
sound.c, sndlib.h) as source files. Add the main program you're interested
in (say sndplay.c), and "Make" the project. When the project is "Run", a
dialog pops up asking for the arguments to the program (in this case the
name of the file to be played, as a quoted string). In the BeOS (Intel),
rename audio.c to audio.cc, then use make (for some reason Be's version of
g++ gets a segfault if the name is audio.c). In Windoze, you can use the C
IDE (a project builder as in the Mac case), or run the compiler from a DOS
shell. In the latter case, (in Watcom C) cl io.c -c -DWINDOZE to create the
object files (io.obj and so on), then

cl sndplay sndplay.obj -DWINDOZE audio.obj io.obj headers.obj sound.obj

or in MS C

cl -c io.c -DWINDOZE
(and so on)
cl sndplay.c -DWINDOZE sndplay.obj audio.obj io.obj headers.obj sound.obj winmm.lib

or in gcc (available via the cygwin project)

gcc -c io.c -DWINDOZE -O2

You can run the program from the DOS shell (sndplay oboe.snd or
./sndplay.exe oboe.snd). Or

make sndplay

To make sndlib into a shared library,

ld -shared io.o headers.o audio.o sound.o -o sndlib.so

(in Linux), or (to include the CLM module),

ld -shared io.o headers.o audio.o sound.o clm.o -o sndlib.so

  ------------------------------------------------------------------------

Current Status

   System    SndSine SndInfo   Audinfo     SndPlay      SndRecord    CLM
 NeXT 68k   ok       ok       ok        ok            ok           ok
 NeXT Intel ok       ok       ok        interruptions runs (*)     ok
 SGI old
 and new AL ok       ok       ok        ok            ok           ok
 OSS (Linux
 et al)     ok       ok       ok        ok            ok           ok

 Be         ok       ok       not       not written   not written  ok
                              written
 Mac        ok       ok       ok        ok            ok           ok
 Windoze    ok       ok       ok        ok            not written  ok
 Sun        ok       ok       ok        ok            interruptionsok
 HPUX       untested untested untested  untested      untested     untried
 LinuxPPC   ok       ok       ok        ok            untested (**)ok
 ALSA       ok       ok       ok        ok            ok           ok

(*) I can't find a microphone.
(**) Last I looked, recording was still not supported in this OS.

   * headers supported read/write
        o NeXT/Sun/DEC/AFsp
        o AIFF/AIFC
        o RIFF (Microsoft wave)
        o IRCAM (old style)
        o NIST-sphere
        o no header
   * headers supported read-only
        o 8SVX (IFF), EBICSF, INRS, ESPS,
        o SPPACK, ADC (OGI), AVR, VOC,
        o Sound Tools, Turtle Beach SMP, SoundFont 2.0,
        o Sound Designer I and II, PSION, MAUD, Kurzweil 2000,
        o Tandy DeskMate, Gravis Ultrasound, ASF, PAF, CSL,
        o Comdisco SPW, Goldwave sample, omf, quicktime
        o Sonic Foundry, SBStudio II, Delusion digital,
        o Digiplayer ST3, Farandole Composer WaveSample,
        o Ultratracker WaveSample, Sample Dump exchange,
        o Yamaha SY85, SY99, and TX16, Covox v8, SPL, AVI

Incomplete: OMF, AVI, ASF, QuickTime, SoundFont 2.0.
Not handled: Esignal, ILS, HTK, DVSM, SoundEdit.
Handled by Snd: Mus10, IEEE text, HCOM, various compression schemes.

Lower Levels

If you'd like to go below the "sound" interface described above, the
following functions are exported from sndlib. You need to remember to call
mus_sound_initialize (or the underlying initializers) before using these
functions (this is normally done for you by the various "sound_" functions).

  int mus_header_read (char *name)
  int mus_header_write (char *name, int type, int in_srate, int in_chans, int loc, int size, int format, char *comment, int len)
  int mus_header_update (char *name, int type, int size, int srate, int format, int chans, int loc)
  int mus_header_writable(int type, int format)

These read and write a sound file's header. The loc parameter is normally 0
(the data location depends on many things -- you'd normally write the
header, then use mus_header_data_location to get the resultant data
location). len is the length (bytes) of comment. mus_header_update is
normally used only to set the file size after the sound has been written.
mus_header_writable returns 1 if the given combination of header type and
data format can be handled by sndlib. If you already have the file
descriptor (as returned by open), the corresponding lower level calls are:

  int mus_header_read_with_fd (int fd)
  int mus_header_write_with_fd (int fd, int type, int in_srate, int in_chans, int loc, int size, int format, char *comment, int len)
  int mus_header_update_with_fd (int fd, int type, int siz)

Once mus_header_read has been called, the data in it can be accessed
through:

  int mus_header_samples (void)              samples
  int mus_header_frames (void)               frames (samples / chans)
  int mus_header_data_location (void)        location of data (bytes)
  int mus_header_chans (void)                channels
  int mus_header_srate (void)                srate
  int mus_header_type (void)                 header type (i.e. aiff, wave, etc)  (see sndlib.h)
  int mus_header_format (void)               data format (see sndlib.h)
  int mus_header_distributed (void)          true if header info is scattered around in the file
  int mus_header_comment_start (void)        comment start location (if any) (bytes)
  int mus_header_comment_end (void)          comment end location
  int mus_header_aux_comment_start (int n)   if multiple comments, nth start location
  int mus_header_aux_comment_end (int n)     if multiple comments, nth end location
  int mus_header_type_specifier (void)       original (header-specific) type ID
  int mus_header_bits_per_sample (void)      sample width in bits
  int mus_header_true_length (void)          true (lseek) file length
  char *mus_header_type_name (int type)      header_type name
  char *mus_data_format_name (int format)    data_format name

Various less useful header fields are accessible: see headers.c or sndlib.h
for details. The next functions handle various IO calls:

  int mus_file_open_read (const char *arg)              open file read-only
  int mus_file_probe (const char *arg)                  return 1 if file exists
  int mus_file_open_write (const char *arg)             open file read-write, creating it if necessary, else truncating
  int mus_file_create (const char *arg)                 create file
  int mus_file_reopen_write (const char *arg)           open file read-write without changing anything
  int mus_file_close (int fd)                           close file
  long mus_file_seek (int tfd, long offset, int origin)
  int mus_file_seek_frame (int tfd, int frame)          go to a specific frame in file
  int mus_file_read (int fd, int beg, int end, int chans, MUS_SAMPLE_TYPE **bufs)
  int mus_file_read_chans (int fd, int beg, int end, int chans, MUS_SAMPLE_TYPE **bufs, MUS_SAMPLE_TYPE *cm)
  int mus_file_read_any (int tfd, int beg, int chans, int nints, MUS_SAMPLE_TYPE **bufs, MUS_SAMPLE_TYPE *cm)
  int mus_file_write_zeros (int tfd, int num)
  int mus_file_write (int tfd, int beg, int end, int chans, MUS_SAMPLE_TYPE **bufs)
  int mus_sound_max_amp (char *ifile, MUS_SAMPLE_TYPE *vals)
  float mus_file_prescaler(int tfd)
  float mus_file_set_prescaler(int tfd, float val)

The prescaler functions are useful in several pathological cases. If you're
reading float data that is extremely soft (i.e. max amp below .001), the
transfer to integer form in sndlib can cause bits to be lost, resulting in
hiss. In this case set the prescaler for the file to 1000.0 or so to get the
data into a more "normal" range. The mus_file_set_prescaler call should come
just after opening the sound file, before trying to read any data.

If you're trying to deal with various data types yourself, the following
functions may be useful; they perform various byte-order-aware type
conversions:

  void mus_bint_to_char (unsigned char *j, int x)
  int mus_char_to_bint (const unsigned char *inp)
  void mus_lint_to_char (unsigned char *j, int x)
  int mus_char_to_lint (const unsigned char *inp)
  int mus_char_to_uninterpreted_int (unsigned char *inp)
  void mus_bfloat_to_char (const unsigned char *j, float x)
  float mus_char_to_bfloat (const unsigned char *inp)
  void mus_lfloat_to_char (unsigned char *j, float x)
  float mus_char_to_lfloat (const unsigned char *inp)
  void mus_bshort_to_char (unsigned char *j, short x)
  short mus_char_to_bshort (const unsigned char *inp)
  void mus_lshort_to_char (unsigned char *j, short x)
  short mus_char_to_lshort (const unsigned char *inp)
  void mus_ubshort_to_char (unsigned char *j, unsigned short x)
  unsigned short mus_char_to_ubshort (const unsigned char *inp)
  void mus_ulshort_to_char (unsigned char *j, unsigned short x)
  unsigned short mus_char_to_ulshort (unsigned char *inp)
  double mus_char_to_ldouble (const unsigned char *inp)
  double mus_char_to_bdouble (const unsigned char *inp)
  void mus_bdouble_to_char (unsigned char *j, double x)
  void mus_ldouble_to_char (unsigned char *j, double x)
  unsigned int mus_char_to_ubint (const unsigned char *inp)
  unsigned int mus_char_to_ulint (const unsigned char *inp)

Finally, a couple functions are provided to read and write sound files to
and from arrays:

  int mus_file_to_array (const char *filename, int chan, int start, int samples, MUS_SAMPLE_TYPE *array)
  int mus_array_to_file (const char *filename, MUS_SAMPLE_TYPE *ddata, int len, int srate, int channels)

Sndlib and Guile

Much of sndlib is accessible at run time in any program that has Guile; the
modules sndlib2scm and clm2scm tie most of the library into Scheme making it
possible to call the library functions from Guile. The documentation is
scattered around, unfortunately: the clm side is in clm.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.

  mus-next mus-aifc mus-riff mus-nist mus-raw mus-ircam mus-aiff

  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-audio-default mus-audio-duplex-default mus-audio-line-out mus-audio-line-in
  mus-audio-microphone mus-audio-speakers mus-audio-dac-out mus-audio-adat-in
  mus-audio-aes-in mus-audio-digital-in mus-audio-digital-out mus-audio-adat-out
  mus-audio-aes-out mus-audio-dac-filter mus-audio-mixer mus-audio-line1
  mus-audio-line2 mus-audio-line3 mus-audio-aux-input mus-audio-cd mus-audio-aux-output
  mus-audio-spdif-in mus-audio-spdif-out

  mus-audio-amp mus-audio-srate mus-audio-channel mus-audio-format mus-audio-port
  mus-audio-imix mus-audio-igain mus-audio-reclev mus-audio-pcm mus-audio-pcm2
  mus-audio-ogain mus-audio-line mus-audio-synth mus-audio-bass mus-audio-treble
  mus-audio-direction mus-audio-samples-per-channel

  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-max-amp(filename)              returns a vector 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
  data-format-bytes-per-sample (format)    bytes per sample

  mus-audio-error ()                       returns error code indicated by preceding audio call
  mus-audio-error-name(err)                string decription of error code
  mus-audio-describe ()                    describe audio hardware state
  mus-audio-report ()                      return audio hardware state as a string
  mus-audio-set-oss-buffers (num size)     in Linux (OSS) sets the number and size of the OSS "fragments"
  mus-audio-sun-outputs (speaker headphones line-out) On the Sun, cause output to go to the chosen 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 accomodate the read
  mus-sound-write (fd beg end chans sdata) write data to sound file fd
  mus-sound-seek (fd offset origin)        complicated -- see mus_sound_seek above
  mus-sound-seek-frame (fd frame)          move to frame in sound file fd
  mus-file-set-data-clipped (fd clipped)   set whether output is clipped

  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-audio-mixer-read (device field channel vals)
                                           read current state of device's field -- see mus_audio_mixer_read above.
  mus-audio-mixer-write (device field channel vals)
                                           write new state for device's field -- see mus_audio_mixer_write above.
  mus-audio-systems ()                     returns how many separate "systems" (soundcards) it can find.  To specify
                                           a particular system in the "device" parameters, add (ash system 16) to the device.
  mus-audio-save ()                        write current audio state to .mixer or whatever
  mus-audio-restore ()                     read previously stored audio state

  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))))

;;; here we get the microphone volume, then set it to .5
  (define vals (make-vector 32))
  (mus-audio-mixer-read mus-audio-microphone mus-audio-amp 0 vals)
  (vector-ref vals 0)
  (vector-set! vals 0 .5)
  (mus-audio-mixer-write mus-audio-microphone mus-audio-amp 0 vals)

;;; this function plays a sound (we're assuming that we can play 16-bit linear little-endian data)
(define play-sound
  (lambda (file)
    (let* ((sound-fd (mus-sound-open-input file))
           (chans (mus-sound-chans file))
           (frames (mus-sound-frames file))
           (bufsize 256)
           (data (make-sound-data chans bufsize))
           (bytes (* bufsize chans 2)))
      (mus-sound-read sound-fd 0 (1- bufsize) chans data)
      (let ((audio-fd (mus-audio-open-output mus-audio-default (mus-sound-srate file) chans mus-lshort bytes)))
        (do ((i 0 (+ i bufsize)))
            ((>= i frames))
          (mus-audio-write audio-fd data bufsize)
          (mus-sound-read sound-fd 0 (1- bufsize) chans data))
        (mus-sound-close-input sound-fd)
        (mus-audio-close audio-fd)))))
