SndLib
Bill Schottstaedt (bil@ccrma.stanford.edu)
Contents
Introduction
Headers
Data
Hardware
Music V
Examples
SndInfo
SndPlay
SndRecord
AudInfo
SndSine
clmosc
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, MkLinux/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 6.0, I changed most of the exported names to use the prefix "mus"
or "SNDLIB" (to be more in line with the Gnu standard); see old-sndlib.h for
backwards compatibility.
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 sound_samples (char *arg) /* samples of sound according to header (can be incorrect) */
int sound_frames (char *arg) /* samples per channel */
float sound_duration (char *arg)
int sound_datum_size (char *arg) /* bytes per sample */
int sound_data_location (char *arg) /* location of first sample (bytes) */
int sound_chans (char *arg) /* number of channels (samples are interleaved) */
int sound_srate (char *arg) /* sampling rate */
int sound_header_type (char *arg) /* header type (aiff etc) */
int sound_data_format (char *arg) /* data format (alaw etc) */
int sound_original_format (char *arg) /* unmodified data format specifier */
char *sound_comment (char *arg) /* comment if any */
int sound_comment_start (char *arg) /* comment start (bytes) if any */
int sound_comment_end (char *arg) /* comment end (bytes) */
int sound_length (char *arg) /* true file length (for error checks) */
int sound_fact_samples (char *arg) /* compression scheme data */
int sound_distributed (char *arg) /* is header scattered around in sound file */
int sound_write_date (char *arg) /* bare (uninterpreted) file write date */
int sound_type_specifier (char *arg) /* original header type identifier */
int sound_align (char *arg) /* more compression data */
int sound_bits_per_sample(char *arg) /* bits per sample */
int sound_bytes_per_sample(int format) /* bytes per sample */
int sound_max_amp(char *arg, int *vals)/* return list of max-amp sample pairs */
int sound_aiff_p(char *arg) /* is it an old-style AIFF file (not AIFC) */
void initialize_sndlib(void) /* initialize everything */
int sound_aiff_p(char *arg) /* if sound's header actually an old-style AIFF (not AIFC) header */
The following can be used to provide user-understandable descriptions of the
header type and the data format:
char *sound_type_name(int type) /* "AIFF" etc */
char *sound_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 audio_error(void) /* returns error code indicated by preceding audio call */
char *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 open_sound_input (char *arg)
int open_sound_output (char *arg, int srate, int chans, int data_format, int header_type, char *comment)
int reopen_sound_output (char *arg, int type, int format, int data_loc)
int close_sound_input (int fd)
int close_sound_output (int fd, int bytes_of_data)
int read_sound (int fd, int beg, int end, int chans, int **bufs)
int write_sound (int fd, int beg, int end, int chans, int **bufs)
int seek_sound (int fd, long offset, int origin)
int seek_sound_frame (int fd, int frame)
int mus_float_sound(char *charbuf, int samps, int charbuf_format, float *buffer)
open_sound_input opens arg for reading. Most standard uncompressed formats
are readable. This function returns the associated file number, or -1 upon
failure.
close_sound_input closes an open sound file. Its argument is the integer
returned by open_sound_input.
open_sound_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: SNDLIB_16_LINEAR, 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 AIFF_sound_file, RIFF_sound_file ('wave'),
NeXT_sound_file, and IRCAM_sound_file), 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.
close_sound_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 open_sound_output.
read_sound 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 read_sound from its
index beg to end with zero padding if necessary. See the sndplay example
below if this is not obvious.
write_sound writes data to the file indicated by fd, starting for each of
chans channels in obufs at beg and ending at end.
seek_sound 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 seek_sound will compensate if its actually 32-bit
floats or whatever. Since this is impossible to understand, there's also
seek_sound_frame which moves to the indicated frame.
mus_float_sound takes a buffer full of sound data in some format
(charbuf_format and returns the data as a buffer full of (unscaled) floats.
Hardware
The following functions provide access to audio harware. If an error occurs,
they return -1, and the audio_error functions can be used to find out what
went wrong.
int initialize_audio(void)
void save_audio_state(void)
void restore_audio_state(void)
void describe_audio_state(void)
char *report_audio_state(void)
int open_audio_output(int dev, int srate, int chans, int format, int size)
int open_audio_input(int dev, int srate, int chans, int format, int size)
int write_audio(int line, char *buf, int bytes)
int close_audio(int line)
int read_audio(int line, char *buf, int bytes)
int read_audio_state(int dev, int field, int chan, float *val)
int write_audio_state(int dev, int field, int chan, float *val)
int audio_systems(void)
char *audio_system_name(int system)
void setup_dsps(int cards, int *dsps, int *mixers) /* OSS only */
initialize_audio takes care of any necessary intialization.
save_audio_state saves the current audio hardware state.
restore_audio_state restores the audio hardware to the last saved state.
describe_audio_state prints to stdout a description of the current state of
the audio hardware. report_audio_state returns the same description as a
string.
audio_systems returns the number of separate and complete audio systems
(soundcards essentially) that are available. audio_system_name returns some
user-recognizable name for the given card.
open_audio_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 SNDLIB_DEFAULT_DEVICE). 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
SNDLIB_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. SNDLIB_AUDIO_SYSTEM(n) refers to the nth such card, so
(SNDLIB_DAC_DEVICE | SNDLIB_AUDIO_SYSTEM(1)) is the 2nd card's dac (the
default is system 0, the first card).
open_audio_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.
close_audio closes the port (input or output) associated with line.
read_audio reads sound data from line. The incoming bytes bytes of data are
placed in buf. If no error was returned from open_audio_input, the data is
in the format requested by that function with channels interleaved.
write_audio 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
open_audio_output with channels interleaved.
read_audio_state and write_audio_state 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 SNDLIB_MICROPHONE_DEVICE, and its
hardware gain setting (if any) is called the SNDLIB_AMP_FIELD. 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;
write_audio_state(SNDLIB_MICROPHONE_DEVICE,SNDLIB_AMP_FIELD,0,vals);
Similarly
read_audio_state(SNDLIB_MICROPHONE_DEVICE,SNDLIB_AMP_FIELD,0,vals);
amp=vals[0];
returns the current gain in the float array vals. read_audio_state 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 SNDLIB_CANT_READ or SNDLIB_CANT_WRITE. If an
error occurs during the requested operation, -1 is returned, and
SNDLIB_AUDIO_ERROR is set to SNDLIB_READ_ERROR or SNDLIB_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 (SNDLIB_SRATE_NOT_AVAILABLE,
SNDLIB_FORMAT_NOT_AVAILABLE, and so on).
Systems
Each separate sound card is called a system, accessible via the device
argument through the macro SNDLIB_AUDIO_SYSTEM(n). The count starts at 0
which is the default. The function 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];
read_audio_state(SNDLIB_AUDIO_SYSTEM(0),SNDLIB_DEVICE_FIELD,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 (SNDLIB_LINE_IN_DEVICE for example). Two special devices are
SNDLIB_MIXER_DEVICE and SNDLIB_DAC_FILTER_DEVICE. 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 read-audio-state and write-audio-state selects one
aspect of the given card's devices' controls. The simplest operations
involve SNDLIB_AMP_FIELD and SNDLIB_SRATE_FIELD. 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 read-audio-state how big a
returned list can get. A brief description of the fields:
SNDLIB_AMP_FIELD gain or volume control (0.0 to 1.0)
SNDLIB_SRATE_FIELD sampling rate
SNDLIB_CHANNEL_FIELD active channels
SNDLIB_BASS_FIELD, SNDLIB_TREBLE_FIELD mixer's tone control
SNDLIB_LINE_FIELD mixer's line-in gain control
SNDLIB_MIC_FIELD mixer's microphone gain control
similarly for SNDLIB_IMIX_FIELD, SNDLIB_IGAIN_FIELD,
SNDLIB_RECLEV_FIELD, SNDLIB_PCM_FIELD, SNDLIB_PCM2_FIELD,
SNDLIB_OGAIN_FIELD, SNDLIB_LINE1_FIELD,
SNDLIB_LINE2_FIELD, SNDLIB_LINE3_FIELD, SNDLIB_SYNTH_FIELD
SNDLIB_FORMAT_FIELD return list of usable sound formats (e.g. SNDLIB_16_LINEAR)
SNDLIB_DEVICE_FIELD return list of available devices (e.g. SNDLIB_MICROPHONE_DEVICE)
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 and examp.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];
initialize_sndlib();
fd = mus_open_read(argv[1]); /* see if it exists */
if (fd != -1)
{
close(fd);
date = sound_write_date(argv[1]);
srate = sound_srate(argv[1]);
chans = sound_chans(argv[1]);
samples = sound_samples(argv[1]);
comment = 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",
sound_type_name(sound_header_type(argv[1])),
sound_format_name(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;
int **bufs;
short *obuf;
initialize_sndlib();
fd = open_sound_input(argv[1]);
if (fd != -1)
{
chans = sound_chans(argv[1]);
srate = sound_srate(argv[1]);
frames = sound_frames(argv[1]);
outbytes = BUFFER_SIZE * chans * 2;
bufs = (int **)calloc(chans,sizeof(int *));
for (i=0;i<chans;i++) bufs[i] = (int *)calloc(BUFFER_SIZE,sizeof(int));
obuf = (short *)calloc(BUFFER_SIZE * chans,sizeof(short));
afd = open_audio_output(SNDLIB_DEFAULT_DEVICE,srate,chans,SNDLIB_COMPATIBLE_FORMAT,outbytes);
if (afd != -1)
{
for (i=0;i<frames;i+=BUFFER_SIZE)
{
read_sound(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] = bufs[n][k];
write_audio(afd,(char *)obuf,outbytes);
}
close_audio(afd);
}
close_sound_input(fd);
for (i=0;i<chans;i++) free(bufs[i]);
free(bufs);
free(obuf);
}
else
fprintf(stderr,"%s: %s ",argv[1],audio_error_name(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 read_audio_state 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;
initialize_sndlib();
fd = open_sound_output(argv[1],22050,1,SNDLIB_16_LINEAR,NeXT_sound_file,"created by sndrecord");
if (fd != -1)
{
ibuf = (short *)calloc(BUFFER_SIZE,sizeof(short));
afd = open_audio_input(SNDLIB_MICROPHONE_DEVICE,22050,1,SNDLIB_16_LINEAR,BUFFER_SIZE);
if (afd != -1)
{
for (i=0;i<10;i++) /* grab 10 buffers of input */
{
err = read_audio(afd,(char *)ibuf,BUFFER_SIZE*2);
if (err != SNDLIB_NO_ERROR) {fprintf(stderr,audio_error_name(audio_error())); break;}
write(fd,ibuf,BUFFER_SIZE*2);
}
close_audio(afd);
}
else
fprintf(stderr,audio_error_name(audio_error()));
close_sound_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[])
{
initialize_sndlib();
describe_audio_state();
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;
int *obuf[1];
initialize_sndlib();
fd = open_sound_output(argv[1],22050,1,SNDLIB_16_LINEAR,NeXT_sound_file,"created by sndsine");
if (fd != -1)
{
frames = 22050;
phase = 0.0;
incr = 2*PI*440.0/22050.0;
obuf[0] = (int *)calloc(BUFFER_SIZE,sizeof(int));
k=0;
for (i=0;i<frames;i++)
{
obuf[0][k] = (int)(3276.8 * sin(phase)); /* amp = .1 */
phase += incr;
k++;
if (k == BUFFER_SIZE)
{
write_sound(fd,0,BUFFER_SIZE-1,1,obuf);
k=0;
}
}
if (k>0) write_sound(fd,0,k-1,1,obuf);
close_sound_output(fd,22050*mus_format2bytes(SNDLIB_16_LINEAR));
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. (Compile clm.c with -DHAVE_SNDLIB=1).
int main(int argc, char *argv[])
{
int i;
mus_any *osc,*op;
initialize_sndlib();
init_mus_module();
osc = mus_make_oscil(440.0,0.0);
op = mus_make_file_output("test.snd",22050,1,SNDLIB_16_LINEAR,NeXT_sound_file,"created by clmosc");
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;
initialize_sndlib();
init_mus_module();
op = mus_make_file_output("test.snd",22050,1,SNDLIB_16_LINEAR,NeXT_sound_file,"created by clmosc");
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
where clm.o was compiled with -DHAVE_SNDLIB.
------------------------------------------------------------------------
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 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). On a Be, you can either build a project or use a
makefile. The C compiler's name is mwcc. The tricky part here is that you
have to find and include explicitly the Be audio library, libmedia.so --
look first in beos/system/lib. 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 interruptionsruns (*) untried
SGI old and new
AL ok ok ok ok ok ok
OSS (Linux et
al) ok ok ok ok ok ok
Be ok ok ok ok ok untried
Mac ok ok ok ok ok ok
Windoze ok ok ok ok not ok
written
Sun ok ok ok ok runs (*) ok
HPUX untested untested untested untested untested untried
MkLinux/LinuxPPCok ok ok ok untested ok
(**)
ALSA untested untested untested untested untested untested
(*) I can't find a microphone.
(**) Last I looked, recording was still not supported in this OS.
headers supported read/write
NeXT/Sun/DEC/AFsp
AIFF/AIFC
RIFF (Microsoft wave)
IRCAM (old style)
NIST-sphere
no header
headers supported read-only
8SVX (IFF), IRCAM Vax float, EBICSF, INRS, ESPS,
SPPACK, ADC (OGI), AVR, VOC,
Sound Tools, Turtle Beach SMP, SoundFont 2.0,
Sound Designer I and II, PSION, MAUD, Kurzweil 2000,
Tandy DeskMate, Gravis Ultrasound, ASF,
Comdisco SPW, Goldwave sample, omf, quicktime
Sonic Foundry, SBStudio II, Delusion digital,
Digiplayer ST3, Farandole Composer WaveSample,
Ultratracker WaveSample, Sample Dump exchange,
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
sndlib_initialize (or the underlying initializers) before using these
functions (this is normally done for you by the various "sound_" functions).
int mus_read_header (char *name)
int mus_write_header (char *name, int type, int in_srate, int in_chans, int loc, int size, int format, char *comment, int len)
int mus_update_header (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_update_header 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_read_header_with_fd (int fd)
int mus_write_header_with_fd (int fd, int type, int in_srate, int in_chans, int loc, int size, int format, char *comment, int len)
int mus_update_header_with_fd (int fd, int type, int siz)
Once mus_read_header 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_true_file_length (void) true (lseek) file length
int mus_header_format2bytes (void) sample width in bytes
int mus_header_aiff_p(void) is header actually old-style AIFF, not AIFC
char *mus_header_type2string (int type) sound_type_name
char *mus_header_data_format2string (int format) sound_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_open_read (char *arg) open file read-only
int mus_probe_file (char *arg) return 1 if file exists
int mus_open_write (char *arg) open file read-write, creating it if necessary, else truncating
int mus_create (char *arg) create file
int mus_reopen_write (char *arg) open file read-write without changing anything
int mus_close (int fd) close file
long mus_seek (int tfd, long offset, int origin)
int mus_seek_frame (int tfd, int frame) go to a specific frame in file
int mus_read (int fd, int beg, int end, int chans, int **bufs)
int mus_read_chans (int fd, int beg, int end, int chans, int **bufs, int *cm)
int mus_read_any (int tfd, int beg, int chans, int nints, int **bufs, int *cm)
int mus_write_zeros (int tfd, int num)
int mus_write (int tfd, int beg, int end, int chans, int **bufs)
int mus_float_sound (char *charbuf, int samps, int charbuf_format, float *buffer)
int mus_unshort_sound (short *in_buf, int samps, int new_format, char *out_buf)
int sound_max_amp (char *ifile, int *vals)
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_set_big_endian_int (unsigned char *j, int x)
int mus_big_endian_int (unsigned char *inp)
void mus_set_little_endian_int (unsigned char *j, int x)
int mus_little_endian_int (unsigned char *inp)
int mus_uninterpreted_int (unsigned char *inp)
void mus_set_big_endian_float (unsigned char *j, float x)
float mus_big_endian_float (unsigned char *inp)
void mus_set_little_endian_float (unsigned char *j, float x)
float mus_little_endian_float (unsigned char *inp)
void mus_set_big_endian_short (unsigned char *j, short x)
short mus_big_endian_short (unsigned char *inp)
void mus_set_little_endian_short (unsigned char *j, short x)
short mus_little_endian_short (unsigned char *inp)
void mus_set_big_endian_unsigned_short (unsigned char *j, unsigned short x)
unsigned short mus_big_endian_unsigned_short (unsigned char *inp)
void mus_set_little_endian_unsigned_short (unsigned char *j, unsigned short x)
unsigned short mus_little_endian_unsigned_short (unsigned char *inp)
double mus_little_endian_double (unsigned char *inp)
double mus_big_endian_double (unsigned char *inp)
void mus_set_big_endian_double (unsigned char *j, double x)
void mus_set_little_endian_double (unsigned char *j, double x)
unsigned int mus_big_endian_unsigned_int (unsigned char *inp)
unsigned int mus_little_endian_unsigned_int (unsigned char *inp)
Finally, a couple functions are provided to read and write sound files to
and from arrays:
int mus_file2array (char *filename, int chan, int start, int samples, int *array)
int mus_array2file (char *filename, int *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.
snd-16-linear snd-16-linear-little-endian snd-24-linear snd-24-linear-little-endian
snd-32-float snd-32-float-little-endian snd-32-linear snd-32-linear-little-endian
snd-64-double snd-64-double-little-endian snd-8-alaw snd-8-linear
snd-8-mulaw snd-8-unsigned snd-16-unsigned snd-16-unsigned-little-endian
next-sound-file nist-sound-file aiff-sound-file ircam-sound-file raw-sound-file riff-sound-file
sndlib-default-device sndlib-read-write-device sndlib-line-out-device
sndlib-line-in-device sndlib-microphone-device sndlib-speakers-device
sndlib-dac-out-device sndlib-adat-in-device sndlib-aes-in-device
sndlib-digital-in-device sndlib-digital-out-device sndlib-adat-out-device
sndlib-aes-out-device sndlib-dac-filter-device sndlib-mixer-device
sndlib-line1-device sndlib-line2-device sndlib-line3-device
sndlib-aux-input-device sndlib-cd-in-device sndlib-aux-output-device
sndlib-spdif-in-device sndlib-spdif-out-device
sndlib-amp-field sndlib-srate-field sndlib-channel-field
sndlib-format-field sndlib-device-field sndlib-imix-field
sndlib-igain-field sndlib-reclev-field sndlib-pcm-field
sndlib-pcm2-field sndlib-ogain-field sndlib-line-field
sndlib-mic-field sndlib-line1-field sndlib-line2-field
sndlib-line3-field sndlib-synth-field sndlib-bass-field
sndlib-treble-field sndlib-cd-field
sound-samples (filename) samples of sound according to header (can be incorrect)
sound-frames (filename) frames of sound according to header (can be incorrect)
sound-duration (filename) duration of sound in seconds
sound-datum-size (filename) bytes per sample
sound-data-location (filename) location of first sample (bytes)
sound-chans (filename) number of channels (samples are interleaved)
sound-srate (filename) sampling rate
sound-header-type (filename) header type (e.g. aiff-sound-file)
sound-data-format(filename) data format (e.g. 16-linear)
sound-length (filename) true file length (bytes)
sound-type-specifier (filename) original header type identifier
sound-max-amp(filename) returns a vector of max amps and locations thereof
sound-type-name (type) e.g. "AIFF"
sound-format-name (format) e.g. "16-bit big endian linear"
sound-comment (filename) header comment, if any
sound-bytes-per-sample (format) bytes per sample
audio-error () returns error code indicated by preceding audio call
audio-error-name(err) string decription of error code
describe-audio () describe audio hardware state
report-audio-state() return audio hardware state as a string
set-oss-buffers (num size) in Linux (OSS) sets the number and size of the OSS "fragments"
audio-outputs(speaker, headphones, line-out) On the Sun, cause output to go to the chosen devices
open-sound-input (filename) open filename (a sound file) returning an integer ("fd" below)
open-sound-output (filename srate chans data-format header-type comment)
create a new sound file with the indicated attributes, return "fd"
reopen-sound-output (filename chans data-format header-type data-location)
reopen (without disturbing) filename, ready to be written
close-sound-input (fd) close sound file
close-sound-output (fd bytes) close sound file and update its length indication, if any
read-sound (fd beg end chans sdata) read data from sound file fd from frame beg to end
sdata is a sound-data object that should be able to accomodate the read
write-sound (fd beg end chans sdata) write data to sound file fd
seek-sound (fd offset origin) complicated -- see seek_sound above
seek-sound-frame (fd frame) move to frame in sound file fd
open-audio-output (device srate chans format bufsize)
open audio port device ready for output with the indicated attributes
open-audio-input (device srate chans format bufsize)
open audio port device ready for input with the indicated attributes
write-audio (line sdata frames) write frames of data from sound-data object sdata to port line
read-audio (line sdata frames) read frames of data into sound-data object sdata from port line
close-audio (line) close audio port line
read-audio-state (device field channel vals)
read current state of device's field -- see read_audio_state above.
write-audio-state (device field channel vals)
write new state for device's field -- see write_audio_state above.
audio-systems () returns how many separate "systems" (soundcards) it can find
save-audio-state () write current audio state to .mixer or whatever
restore-audio-state () 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 (sound-chans file))
", srate: " (number->string (sound-srate file))
", " (sound-type-name (sound-header-type file))
", " (sound-format-name (sound-data-format file))
", len: " (number->string
(/ (sound-samples file)
(* (sound-chans file) (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 (open-sound-input file))
(chans (sound-chans file))
(data (make-sound-data chans 32)))
(read-sound 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)))
(close-sound-input fd)
val))))
;;; here we get the microphone volume, then set it to .5
(define vals (make-vector 32))
(read-audio-state sndlib-microphone-device sndlib-amp-field 0 vals)
(vector-ref vals 0)
(vector-set! vals 0 .5)
(write-audio-state sndlib-microphone-device sndlib-amp-field 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 (open-sound-input file))
(chans (sound-chans file))
(frames (sound-frames file))
(bufsize 256)
(data (make-sound-data chans bufsize))
(bytes (* bufsize chans 2)))
(read-sound sound-fd 0 (1- bufsize) chans data)
(let ((audio-fd (open-audio-output sndlib-default-device (sound-srate file) chans snd-16-linear-little-endian bytes)))
(do ((i 0 (+ i bufsize)))
((>= i frames))
(write-audio audio-fd data bufsize)
(read-sound sound-fd 0 (1- bufsize) chans data))
(close-sound-input sound-fd)
(close-audio audio-fd)))))