# 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
- MIDI
- Music V
- Examples
  - SndInfo
  - SndPlay
  - SndRecord
  - AudInfo
  - SndSine
  - clmosc
  - Other Examples
- Current Status
- Extension Languages

## Introduction

sndlib is a collection of sound file and audio hardware handlers written in C and running currently on SGI (either audio library), Sun, OSS or ALSA (Linux and others), Mac OSX, and 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, SGI, and Sun. It could conceivably work elsewhere.

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, Guile, Ruby, or Forth)
- clm.c and clm.h (Music V implementation)
- clm2xen.c, vct.c and vct.h (tie clm.c into s7, Guile, Ruby, or Forth)
- xen.h, xen.c, xen.html (the embedded language support)
- run.c (the run macro to speed up scheme computations)
- midi.c (primitive MIDI support)

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 MIDI functions with "mus_midi_", 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)
      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)

mus_audio_initialize takes care of any necessary initialization.

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_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. 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 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.

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 or something goes wrong, -1 is returned.

#### 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)

## MIDI

There is some simple, very low-level MIDI support in sndlib.

     int mus_midi_open_read(const char *name)
     int mus_midi_open_write(const char *name)
     int mus_midi_close(int line)
     int mus_midi_read(int line, unsigned char *buffer, int bytes)
     int mus_midi_write(int line, unsigned char *buffer, int bytes)
     char *mus_midi_device_name(int sysdev)
     char *mus_midi_describe(void)

These procedures remove all structure imposed by the available MIDI drivers, presenting each such device as if it were a sort of file. mus_midi_open_read and mus_midi_open_write open a given device (its name can be found by calling mus_midi_device_name with an argument of 0), returning an integer that refers to that port in subsequent calls on mus_midi_read, mus_midi_write, and mus_midi_close. No attempt is made to interpret the data, or time its output, etc. mus_midi_describe returns a description of the available MIDI hardware (its result should be freed, whereas the result of mus_midi_device_name should not be freed). The "int" result of each of the functions is -1 if some error occurred. The sysdev argument to mus_midi_device_name is a sndlib-style integer packing the card number ("system" in the left half, and the device number in the right). On the ALSA system I'm testing at the moment, the MIDI device is apparently "hw:1,0" or "/dev/snd/midiC1D0"; this maps into sndlib as card 1, device 0; the corresponding "sysdev" number is 1 << 16. The corresponding functions are:

      mus-midi-open-read name
      mus-midi-open-write name
      mus-midi-close md
      mus-midi-device-name sys-dev
      mus-midi-describe
      mus-midi-read md bytes
      mus-midi-write md bytes

mus-midi-read and mus-midi-write return a list of the bytes read or written (empty if none), or #f if something went wrong.

## MusicV

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 Guile, 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-11.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-11.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;
        init_mus_module();
        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, ...)

Before using any of these functions, call init_mus_module. 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);
    }

### 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 mus_file_read_buffer to turn that data into a stream of floats. You'd also provide some way to turn the thing off. (sndrecord.c)

    int main(int argc, char *argv[])
    {
      int fd, afd, i, err;
      short *ibuf;
      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_NO_ERROR) break;
    	      write(fd, ibuf, BUFFER_SIZE * 2);
    	    }
    	  mus_audio_close(afd);
    	}
          mus_sound_close_output(fd, BUFFER_SIZE * 10 * 2);
          free(ibuf);
        }
      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_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();
      init_mus_module();
      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, ampfuzz = 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 *osc = NULL, *op = NULL;
      mus_sound_initialize();
      init_mus_module();
      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.

## Current Status

System | SndSine | SndInfo | Audinfo | SndPlay | SndRecord | CLM | MIDI
--- | --- | --- | --- | --- | --- | --- | ---
SGI old and new AL | ok | ok | ok | ok | ok | ok | partly tested
OSS (Linux et al) | ok | ok | ok | ok | ok | ok | untested
Windoze | ok | ok | ok | ok | not written | ok | not written
Sun | ok | ok | ok | ok | ok | ok | not written
LinuxPPC | ok | ok | ok | ok | untested | ok | untested
ALSA | ok | ok | ok | ok | ok | ok | partly tested
Mac OSX | ok | ok | ok | ok | ok | ok | untested
NetBSD | ok | ok | ok | ok | ok | ok | untested
ESD | ok | ok | ok | ok |  | ok | untested
HPUX | ok | ok | ok | ok |  | ok | untested

read/write (many data formats)

        NeXT/Sun/DEC/AFsp
        AIFF/AIFC
        RIFF (Microsoft wave)
        RF64
        IRCAM (old style)
        NIST-sphere
        CAFF
        no header ("raw")

read-only (in selected data formats)

        8SVX (IFF), EBICSF, INRS, ESPS, SPPACK, ADC (OGI), AVR, VOC, PVF, NVF,
        Sound Tools, Turtle Beach SMP, SoundFont 2.0, Sound Designer I, PSION, MAUD, Kurzweil 2000,
        Gravis Ultrasound, ASF, PAF, CSL, Comdisco SPW, Goldwave sample, omf, quicktime, sox,
        Sonic Foundry, SBStudio II, Delusion digital, Digiplayer ST3, Farandole Composer WaveSample,
        Ultratracker WaveSample, Sample Dump exchange, Yamaha SY85, SY99, and TX16, Covox v8, AVI,
        Impulse tracker, Korg, Akai, Turtle Beach, Matlab-5

automatically translated to a readable format

        IEEE text, Mus10, SAM 16-bit (modes 1 and 4), AVI, NIST shortpack, HCOM, Intel,
        IBM, and Oki (Dialogic) ADPCM, G721, G723_24, G723_40, MIDI sample dump, Ogg, Speex,
        Flac, Midi, Mpeg, Shorten, Wavepack, tta (via external programs)

## Extension Languages

Much of sndlib is accessible at run time in any program that has one of the languages supported by the xen package (Guile, 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 (Guile, 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-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-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-describe ()                    describe audio hardware state
      mus-audio-report ()                      return audio hardware state as a string
      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-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-oss-set-buffers (num size)           in Linux (OSS) sets the number and size of the OSS "fragments"
      mus-sun-set-outputs (speaker headphones line-out) On the Sun, cause output to go to the chosen devices
      mus-netbsd-set-outputs (speaker headphones line-out) On netBSD, cause output to go to the chosen devices

      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-vct 32))
      (mus-audio-mixer-read mus-audio-microphone mus-audio-amp 0 vals)
      (vct-ref vals 0)
      (vct-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 (- bufsize 1) 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 (- bufsize 1) chans data))
    	(mus-sound-close-input sound-fd)
    	(mus-audio-close audio-fd)))))

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