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 (NeXTStep and OpenStep), Sun, Be, OSS (Linux and others), Mac, HPUX, MkLinux, 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), and sound.c (provide slightly higher level access to the preceding files). The overall header file is sndlib.h. (The stub file lispcall.c takes care of a function used in C-to-Lisp communication).
Sound files have built-in descriptors known as headers (we're in computer land, so no header is considered a kind of header). 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_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 bytes_per_sample(int format) /* bytes per sample */
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.
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 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) void 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: snd_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.
float_sound takes a buffer full of sound data in some format (charbuf_format and returns the data as a buffer full of floats.
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 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 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. AUDIO_SYSTEM(n) refers to the nth such card, so (DAC_DEVICE | 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 MICROPHONE_DEVICE, and its hardware gain setting (if any) is called the 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(MICROPHONE_DEVICE,AMP_FIELD,0,vals);
Similarly
read_audio_state(MICROPHONE_DEVICE,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 AUDIO_ERROR is set to CANT_READ or CANT_WRITE. If an error occurs during the requested operation, -1 is returned, and AUDIO_ERROR is set to READ_ERROR or WRITE_ERROR. If some operation cannot be performed on the current hardware, -1 is returned and AUDIO_ERROR tries to indicate what portion of the requested operation is impossible (SRATE_NOT_AVAILABLE, FORMAT_NOT_AVAILABLE, and so on).
Each separate sound card is called a system, accessible via the device argument through the macro 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).
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(AUDIO_SYSTEM(0),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 (LINE_IN_DEVICE for example). Two special devices are MIXER_DEVICE and 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).
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 AMP_FIELD and 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:
AMP_FIELD gain or volume control (0.0 to 1.0)
SRATE_FIELD sampling rate
CHANNEL_FIELD active channels
BASS_FIELD, TREBLE_FIELD mixer's tone control
LINE_FIELD mixer's line-in gain control
MIC_FIELD mixer's microphone gain control
similarly for IMIX_FIELD, IGAIN_FIELD, RECLEV_FIELD, PCM_FIELD, PCM2_FIELD,
OGAIN_FIELD, LINE1_FIELD, LINE2_FIELD, LINE3_FIELD, SYNTH_FIELD
FORMAT_FIELD return list of usable sound formats (e.g. snd_16_linear)
DEVICE_FIELD return list of available devices (e.g. MICROPHONE_DEVICE)
Due to minor problems in the OSS (Linux) multi-card support, you sometimes need to set up the map of dsps and mixers by hand. setup_dsps, called after audio_initialize, sets the number of cards and the device numbers of the main dsp (0 for /dev/dsp0), and mixer.
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.
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];
fd = clm_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);
}
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;
fd = open_sound_input(argv[1]);
if (fd != -1)
{
initialize_audio();
chans = sound_chans(argv[1]);
srate = sound_srate(argv[1]);
frames = sound_samples(argv[1])/chans;
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(DEFAULT_DEVICE,srate,chans,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);
}
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;
fd = open_sound_output(argv[1],22050,1,snd_16_linear,NeXT_sound_file,"created by sndrecord");
if (fd != -1)
{
ibuf = (short *)calloc(BUFFER_SIZE,sizeof(short));
afd = open_audio_input(MICROPHONE_DEVICE,22050,1,snd_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 != 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);
}
This program describes the current audio harware state (audinfo.c):
int main(int argc, char *argv[])
{
describe_audio_state();
return(0);
}
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];
fd = open_sound_output(argv[1],22050,1,snd_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*c_snd_datum_size(snd_16_linear));
free(obuf[0]);
}
return(0);
}
The primary impetus for the sound library was the development of Snd and CLM, both of which are freely available.
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 lispcall.c -O -DSGI cc -c sound.c -O -DSGI cc sndplay.c -o sndplay -O -DSGI audio.o io.o headers.o lispcall.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 lispcall.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 lispcall.c -O -DLINUX cc -c sound.c -O -DLINUX cc sndplay.c -o sndplay -O -DLINUX audio.o io.o headers.o lispcall.o sound.o -lm ld -r audio.o io.o headers.o lispcall.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 lispcall.c -O -DNEXT cc -c sound.c -O -DNEXT cc sndplay.c -o sndplay -O -DNEXT audio.o io.o headers.o lispcall.o sound.o ld -r audio.o io.o headers.o lispcall.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 or in Windoze, you need to make a project in CodeWarrior or Watcom or whatever that includes all the basic sndlib .c and .h files (io.c, audio.c headers.c, lispcall.c, sound.c, sndlib.h) as source files. Add the main program you're interested in (say sndplay.c), and "Make" the project. On a Mac, 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 run the program from a DOS shell. 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. The Snd package includes make information for the sndlib examples as well, so if you have Snd, you can say:
make sndplay
and so on.
| System | SndSine | SndInfo | Audinfo | SndPlay | SndRecord |
|---|---|---|---|---|---|
| NeXT 68k | ok | ok | ok | ok | ok |
| NeXT Intel | ok | ok | ok | interruptions | runs (*) |
| SGI old and new AL | ok | ok | ok | ok | ok |
| OSS (Linux et al) | ok | ok | ok | ok | ok |
| Be | ok | ok | ok | ok | ok |
| Mac | ok | ok | ok | ok | ok |
| Windoze | ok | ok | ok | ok | not written |
| Sun | broken | ok | broken | ok | bus error |
| HPUX | untested | untested | untested | untested | untested |
| MkLinux/LinuxPPC | ok | ok | unready | unready | unready |
(*) I can't find a microphone.
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.