long audio_poll(snd_type snd)
{
oss_info dp = get_oss_info(snd);
/* Note that this returns frames while audio_write
returns bytes */
if (snd->write_flag == SND_READ)
return (dp->len / snd_bytes_per_frame(snd));
else
return ((dp->buffersize - dp->len) /
snd_bytes_per_frame(snd));
}
void snd_print_info(snd_type snd)
{
if (snd->device == SND_DEVICE_AUDIO) {
if (snd->write_flag == SND_READ) {
printf("audio input");
} else {
printf("audio output");
}
} else if (snd->device == SND_DEVICE_MEM) {
if (snd->write_flag == SND_READ) {
printf("from memory buffer");
} else {
printf("to memory buffer");
}
} else {
printf("header %s", header_to_string(snd->u.file.header));
}
printf(", channels %d, mode %s, bits %d, srate %g",
snd->format.channels,
snd_mode_to_string(snd->format.mode),
snd->format.bits, snd->format.srate);
if (snd->device == SND_DEVICE_MEM) {
if (snd->write_flag == SND_READ) {
printf("length %d\n", snd->u.mem.buffer_len);
} else {
printf("length %d\n", snd->u.mem.buffer_max);
}
} else if (snd->device == SND_DEVICE_FILE &&
snd->write_flag == SND_READ) {
printf(", length %d\n", (snd->u.file.end_offset -
snd->u.file.byte_offset) / snd_bytes_per_frame(snd));
} else printf("\n");
}
void *audio_descr_build(snd_type snd)
{
int i, j;
buffer_state *bufstate;
double bufsize_in_frames;
bufstate = (buffer_state *) snd_alloc(sizeof(buffer_state));
if (snd->u.audio.protocol == SND_REALTIME){
bufstate->numofbuffers = 3;
}
else {
if (snd->u.audio.protocol != SND_COMPUTEAHEAD)
snd_fail("Invalid protocol identifier. COMPUTEAHEAD method used instead");
bufstate->numofbuffers = 2;
}
if (NULL == (bufstate->whdr =
(LPWAVEHDR)snd_alloc(bufstate->numofbuffers * sizeof(WAVEHDR)))){
snd_fail("Not enough memory, or similar");
snd_free(bufstate);
return NULL;
}
bufsize_in_frames = max(Audio_out_min_buffersize,
snd->format.srate * snd->u.audio.latency);
if (bufsize_in_frames > snd->format.srate * snd->u.audio.latency)
snd->u.audio.latency = bufsize_in_frames / snd->format.srate;
for (i=0; i<bufstate->numofbuffers; i++){
bufstate->whdr[i].dwBufferLength = (long)(bufsize_in_frames + 0.5) * snd_bytes_per_frame(snd);
if (NULL == (bufstate->whdr[i].lpData = (char *) snd_alloc(bufstate->whdr[i].dwBufferLength))){
snd_fail("Not enough memory, or similar");
for (j=0; j<i-1; j++){
snd_free(bufstate->whdr[j].lpData);
}
snd_free(bufstate);
return NULL;
}
bufstate->whdr[i].dwUser = bufstate->whdr[i].dwLoops = 0; //idrk, what's that
bufstate->whdr[i].dwFlags = WHDR_DONE;
}
bufstate->pollee = 0;
bufstate->posinbuffer = 0;
if (snd->u.audio.protocol == SND_REALTIME){
bufstate->sync_buffer = 0;
bufstate->prev_time = timeGetTime();
bufstate->sync_time = bufstate->prev_time - (long)(2 * 1000 * snd->u.audio.latency + .5);
}
return (void *) bufstate;
}
MMRESULT win_wave_open(snd_type snd, UINT devtoopen, HWAVE *hptr)
{
WAVEFORMATEX wfmt;
int bperf;
switch (snd->format.mode) {
case SND_MODE_UNKNOWN:
case SND_MODE_PCM:
/* note: Windows uses signed PCM (PCM) for 16-bit data, and
* unsigned PCM (UPCM) for 8-bit data
*/
if (snd->format.bits != 8) {
wfmt.wFormatTag = WAVE_FORMAT_PCM;
break;
} else return WAVERR_BADFORMAT;
case SND_MODE_ADPCM:
case SND_MODE_ALAW:
case SND_MODE_ULAW:
case SND_MODE_FLOAT:
case SND_MODE_UPCM:
if (snd->format.bits == 8) {
wfmt.wFormatTag = WAVE_FORMAT_PCM;
break;
} else return(WAVERR_BADFORMAT);
default:
return(MMSYSERR_INVALPARAM);
}
wfmt.nChannels = (unsigned short) snd->format.channels;
wfmt.nSamplesPerSec = (long) (snd->format.srate + 0.5);
bperf = snd_bytes_per_frame(snd);
wfmt.nAvgBytesPerSec = (long) (bperf * snd->format.srate + 0.5);
wfmt.nBlockAlign = (unsigned short) bperf;
wfmt.wBitsPerSample = (unsigned short) snd->format.bits;
wfmt.cbSize = 0;
if (snd->write_flag != SND_READ) {
/* Under Windows2000 on an IBM A21p Thinkpad, waveOutOpen will
CRASH (!) if snd->format.srate is 110250, but it actually
worked with 4000, and it worked with 96000, so I will assume
sample rates above 96000 are bad. -RBD 24Jul02
*/
if (wfmt.nSamplesPerSec > 96000) {
return WAVERR_BADFORMAT;
}
/* otherwise, we're ready to open the device */
return waveOutOpen((LPHWAVEOUT) hptr, devtoopen, &wfmt, 0L, 0L,
(DWORD) CALLBACK_NULL);
} else {
return waveInOpen((LPHWAVEIN) hptr, devtoopen, &wfmt, 0L, 0L,
(DWORD) CALLBACK_NULL);
}
}
long audio_write(snd_type snd, void *buffer, long length_in_bytes)
{
oss_info dp = get_oss_info(snd);
pthread_mutex_lock(&dp->mutex);
if (dp->len + length_in_bytes > dp->buffersize)
length_in_bytes = dp->buffersize - dp->len;
copy_to_wrap_buffer(dp, buffer, length_in_bytes);
pthread_mutex_unlock(&dp->mutex);
return (length_in_bytes / snd_bytes_per_frame(snd));
}
/*
* We want to write as soon as space is available so that
* a full sound buffer can be queued up for output. This
* may require transferring only part of buf, so we keep
* track of progress and output whenever space is available.
*/
void write_to_audio(snd_type player, void *buf, long buflen)
{
long rslt;
while (buflen) {
/* this loop is a busy-wait loop! */
rslt = snd_poll(player); /* wait for buffer space */
rslt = min(rslt, buflen);
if (rslt) {
snd_write(player, buf, rslt);
buf = (void *) ((char *) buf +
(rslt * snd_bytes_per_frame(player)));
buflen -= rslt;
}
}
}
long audio_read(snd_type snd, void *buffer, long length_in_bytes)
{
oss_info dp = get_oss_info(snd);
if (length_in_bytes <= 0)
return 0;
pthread_mutex_lock(&dp->mutex);
if (length_in_bytes > dp->len)
length_in_bytes = dp->len;
copy_from_wrap_buffer(dp, buffer, length_in_bytes);
pthread_mutex_unlock(&dp->mutex);
return (length_in_bytes / snd_bytes_per_frame(snd));
}
int audio_open(snd_type snd, long *flags)
{
int format;
int channels;
int rate;
oss_info dp;
const char *device = "/dev/dsp";
pthread_t thread;
snd->u.audio.descriptor = (oss_info) malloc(sizeof(oss_info_struct));
dp = get_oss_info(snd);
if (snd->u.audio.devicename[0] != 0)
device = snd->u.audio.devicename;
if (snd->write_flag == SND_READ) {
/* open audio input */
/* Open /dev/dsp */
dp->audio_fd = open(device, O_RDONLY, 0);
if (dp->audio_fd == -1)
return !SND_SUCCESS;
}
else {
/* open audio output */
/* Open /dev/dsp */
dp->audio_fd = open(device, O_WRONLY, 0);
if (dp->audio_fd == -1)
return !SND_SUCCESS;
}
/* Set format to signed 16-bit little-endian */
format = AFMT_S16_LE;
if (ioctl(dp->audio_fd, SNDCTL_DSP_SETFMT, &format) == -1)
return !SND_SUCCESS;
if (format != AFMT_S16_LE) /* this format is not supported */
return !SND_SUCCESS;
/* Set number of channels */
channels = snd->format.channels;
if (ioctl(dp->audio_fd, SNDCTL_DSP_CHANNELS, &channels) == -1)
return !SND_SUCCESS;
if (channels != snd->format.channels)
return !SND_SUCCESS;
/* Set sampling rate. Must set sampling rate AFTER setting
number of channels. */
rate = (int)(snd->format.srate + 0.5);
if (ioctl(dp->audio_fd, SNDCTL_DSP_SPEED, &rate) == -1)
return !SND_SUCCESS;
if (rate - (int)(snd->format.srate + 0.5) > 100 ||
rate - (int)(snd->format.srate + 0.5) < -100)
return !SND_SUCCESS;
/* finish initialization and start the thread */
if (snd->write_flag == SND_READ) {
/* always buffer 10 seconds of audio for reading
(no reason not to for this implementation */
dp->buffersize = rate * 10.0 * snd_bytes_per_frame(snd);
}
else {
if (snd->u.audio.latency > 0.0)
dp->buffersize = rate * snd->u.audio.latency *
snd_bytes_per_frame(snd);
else
dp->buffersize = rate * 1.0 * snd_bytes_per_frame(snd);
}
dp->buffer = malloc(dp->buffersize);
dp->start = 0;
dp->len = 0;
dp->error_count = 0;
dp->stop_flag = 0;
pthread_mutex_init (&dp->mutex, 0);
if (snd->write_flag == SND_READ) {
if (0 != pthread_create(&dp->thread, NULL,
record_thread, (void *)dp))
return !SND_SUCCESS;
}
else {
if (0 != pthread_create(&dp->thread, NULL,
playback_thread, (void *)dp))
return !SND_SUCCESS;
}
return SND_SUCCESS;
}
long audio_poll(snd_type snd)
{
long availtemp;
buffer_state *cur = ((buffer_state*) snd->u.audio.descriptor);
LPWAVEHDR whdrs = cur->whdr;
int i, test = 0;
int numbuf = cur->numofbuffers;
for (i=0; i<numbuf; i++){
if (whdrs[i].dwFlags & WHDR_DONE){
test++;
}
}
if (snd->u.audio.protocol == SND_REALTIME) {
int last_buf, k;
if (test < numbuf) {
/* there is at least one buffer being processed by the
* device (not "done")
*/
if (test) { /* there is at last one buffer done */
long bufferbytes;
if (whdrs[cur->sync_buffer].dwFlags & WHDR_DONE){
k = (cur->sync_buffer + 1) % numbuf;
while (k != cur->sync_buffer){
if (!(whdrs[k].dwFlags & WHDR_DONE)){
cur->sync_buffer = k;
break;
}
k = (k + 1) % numbuf;
}
cur->sync_time = cur->prev_time;
} else
cur->prev_time = timeGetTime();
availtemp = (long)
(((double)(cur->prev_time - cur->sync_time) *
max_srate_dev / 1000
/*+ 2 * snd->u.audio.latency*/) *
snd->format.srate);
last_buf = (cur->sync_buffer + numbuf - 1) % numbuf;
while (last_buf != cur->pollee) {
availtemp += whdrs[last_buf].dwBufferLength;
last_buf = (last_buf + numbuf - 1) % numbuf;
}
bufferbytes = test * (long) whdrs[0].dwBufferLength;
if (bufferbytes < availtemp * snd_bytes_per_frame(snd)) {
return bufferbytes / snd_bytes_per_frame(snd);
} else {
return availtemp;
}
} else {
/* all buffer already passed to the device, it
* should happen only just because we "overestimate"
* the sample-rate
*/
cur->prev_time = timeGetTime();
cur->sync_time = cur->prev_time;
return 0;
}
} else { //all buffers are done
cur->prev_time = timeGetTime();
cur->sync_time = cur->prev_time;
return ((numbuf - 1) * whdrs[0].dwBufferLength) /
snd_bytes_per_frame(snd);
}
} else { /* protocol SND_COPMUTEAHEAD */
/* here is suppose that all buffer have the same length
* (otherwise, sg like "for(i=0;i<numbuf;i++)
* available+=whdrs[i].dwBufferLength" should be used)
*/
return (test * whdrs[0].dwBufferLength - cur->posinbuffer) /
snd_bytes_per_frame(snd);
}
}
int audio_open(snd_node *n, long *f)
{
buffer_state *data = (buffer_state *)malloc(sizeof(buffer_state));
n->u.audio.descriptor = (void *)data;
OSErr err;
Fixed sampleRateFixed;
data->frameSize = snd_bytes_per_frame(n);
data->bufferSize = (int) (n->format.srate * (double)data->frameSize);
if (n->u.audio.latency > 0.0)
data->bufferSize = (int)(n->format.srate * n->u.audio.latency) * data->frameSize;
/* Calculate sample rate as an unsigned fixed-point number */
if (n->format.srate > 65535.0 ||
n->format.srate < 1.0)
sampleRateFixed = 0xAC440000; /* Fixed for 44100 */
else {
unsigned short numerator = (unsigned short)n->format.srate;
unsigned short denominator = (unsigned short)(65536.0*(n->format.srate - numerator));
sampleRateFixed = (numerator << 16) | denominator;
}
/* Open device for recording or playback, depending on mode selected */
if (n->write_flag == SND_READ) {
/* recording */
short gainControl = 0; /* off */
short numChannels = n->format.channels;
short continuousRecording = 1; /* on */
short playthroughVolume = 0; /* off */
OSType quality = 'cd ';
short sampleSize = 16;
short twos = 0; /* i.e. signed */
OSType compression = 'NONE';
SoundInfoList infoList;
int len, i, x, y, z;
short selected = 0;
char *device;
char *input;
char **buffer;
char oneInput[256];
char *deviceData;
data->recording = 1;
len = strlen(n->u.audio.devicename);
device = new char[len+1];
input = new char[len+1];
strcpy(device, n->u.audio.devicename);
input[0] = 0;
for(i=0; i<len; i++) {
if (device[i] == '\n') {
device[i] = 0;
strcpy(input, &device[i+1]);
i = len;
}
}
err = SPBOpenDevice(c2pstr(device), siWritePermission, &data->refnum);
if (err)
return !SND_SUCCESS;
SPBGetDeviceInfo (data->refnum, siInputAvailable, &infoList);
SPBGetDeviceInfo (data->refnum, siInputSourceNames, &buffer);
deviceData = buffer[0] + 2;
for(int x=0; x<infoList.count; x++) {
int y = *deviceData++;
z = 0;
while(y) {
oneInput[z++] = *deviceData++;
y--;
}
oneInput[z] = 0;
if (!strcmp(oneInput, input))
selected = x;
}
selected++;
SPBSetDeviceInfo (data->refnum, siInputSource, &selected);
SPBSetDeviceInfo(data->refnum, 'qual', &quality);
SPBSetDeviceInfo(data->refnum, 'agc ', &gainControl);
SPBSetDeviceInfo(data->refnum, 'srat', &sampleRateFixed);
SPBSetDeviceInfo(data->refnum, 'plth', &playthroughVolume);
err = SPBSetDeviceInfo(data->refnum, 'ssiz', &sampleSize);
if (err)
return !SND_SUCCESS;
err = SPBSetDeviceInfo(data->refnum, 'chan', &numChannels);
if (err)
return !SND_SUCCESS;
err = SPBSetDeviceInfo(data->refnum, 'cont', &continuousRecording);
if (err)
return !SND_SUCCESS;
err = SPBSetDeviceInfo(data->refnum, 'twos', &twos);
if (err)
return !SND_SUCCESS;
err = SPBSetDeviceInfo(data->refnum, 'comp', &compression);
if (err)
return !SND_SUCCESS;
data->recBuffer = (char *)malloc(data->bufferSize);
data->recqStart = 0;
data->recqEnd = 0;
data->starved = 0;
data->params.inRefNum = data->refnum;
data->params.count = 0; /* data->bufferSize; /* bytes to record */
data->params.milliseconds = 0; /* param will be ignored; use count */
data->params.bufferLength = 0; /* ignore buffer */
data->params.bufferPtr = NULL; /* ignore buffer */
data->params.completionRoutine = NULL;
data->params.userLong = (long)data;
data->params.unused1 = 0;
data->params.interruptRoutine = NewSIInterruptUPP(recordingCallback);
err = SPBRecord(&data->params, true);
if (err)
return !SND_SUCCESS;
}
else {
/* playback */
data->recording = 0;
data->chan = NULL;
////////////////////////
err = SndNewChannel(&data->chan, sampledSynth, 0, NULL);
if (err)
return !SND_SUCCESS;
data->buffer = (char *)malloc(data->bufferSize);
data->nextBufferSize = data->bufferSize * 3;
data->nextBuffer = (char *)malloc(data->nextBufferSize);
if (!data->buffer || !data->nextBuffer)
return !SND_SUCCESS;
data->chan->callBack = NewSndCallBackUPP(playbackCallback);
data->header.samplePtr = data->buffer;
data->header.numChannels = n->format.channels;
data->header.sampleRate = sampleRateFixed;
data->header.loopStart = 0;
data->header.loopEnd = 0;
data->header.encode = cmpSH;
data->header.baseFrequency = kMiddleC;
// data->header.AIFFSampleRate = 0; -- this parameter is unused
data->header.markerChunk = NULL;
data->header.format = kSoundNotCompressed;
data->header.futureUse2 = NULL;
data->header.stateVars = NULL;
data->header.leftOverSamples = NULL;
data->header.compressionID = 0;
data->header.packetSize = 0;
data->header.snthID = 0;
data->header.sampleSize = 16;
data->header.sampleArea[0] = 0;
data->playCmd.cmd = bufferCmd;
data->playCmd.param1 = 0; //unused
data->playCmd.param2 = (long)&data->header;
data->callCmd.cmd = callBackCmd;
data->callCmd.param1 = 0;
data->callCmd.param2 = (long)data;
data->curBuffer = 0;
data->curSize = 0;
data->firstTime = 1;
data->finished = 0;
data->busy = 0;
data->flushing = 0;
}
return SND_SUCCESS;
}
int main(int argc, char *argv[])
{
char *fromfile = NULL;
char *tofile = NULL;
int i;
long flags = 0;
long frames;
long len;
long checksum = 0;
long count = 0;
int format = SND_HEAD_AIFF;
int mode = SND_MODE_PCM;
int channels = 1;
int bits = 16;
double srate = 44100.0;
snd_node from_snd, to_snd;
char from_buf[MAXLEN];
char to_buf[MAXLEN];
long frames_from_len;
long frames_to_len;
long from_len = 0;
long to_len = 0;
for (i = 1; i < argc; i++) {
if (*(argv[i]) != '-') {
if (!fromfile) fromfile = argv[i];
else if (!tofile) tofile = argv[i];
else {
printf("%s: don't know what to do with 3rd file\n", argv[i]);
return 1;
}
} else {
char *flag = argv[i] + 1;
if (*flag == '?') {
printf("convert -- sound file conversion utility\n\n");
printf("usage: convert fromfile tofile -flag1 -flag2 ...\n");
printf(" -fa -- AIFF file format\n");
printf(" -fi -- IRCAM file format\n");
printf(" -fn -- NeXT/Sun file format\n");
printf(" -fw -- Wave file format\n");
printf(" -ep -- PCM\n");
printf(" -em -- u-Law\n");
printf(" -ef -- float\n");
printf(" -eu -- unsigned PCM\n");
printf(" -b1 -- 8-bit\n");
printf(" -b2 -- 16-bit\n");
printf(" -b4 -- 32-bit\n");
printf(" -c1 -- 1 channel, etc.\n");
printf("use 'ada' to get audio input or output\n");
} else if (*flag == 'f') {
switch (flag[1]) {
case 'a':
format = SND_HEAD_AIFF;
break;
case 'i':
format = SND_HEAD_IRCAM;
break;
case 'n':
format = SND_HEAD_NEXT;
break;
case 'w':
format = SND_HEAD_WAVE;
break;
default:
printf("flag %s ignored\n", argv[i]);
break;
}
} else if (*flag == 'e') {
switch (flag[1]) {
case 'p':
mode = SND_MODE_PCM;
break;
case 'm':
mode = SND_MODE_ULAW;
break;
case 'f':
mode = SND_MODE_FLOAT;
break;
case 'u':
mode = SND_MODE_UPCM;
break;
default:
printf("flag %s ignored\n", argv[i]);
break;
}
} else if (*flag == 'b') {
switch (flag[1]) {
case '1':
bits = 8;
break;
case '2':
bits = 16;
break;
case '4':
bits = 32;
break;
default:
printf("flag %s ignored\n", argv[i]);
break;
}
} else if (*flag == 'r') {
switch (flag[1]) {
case '4':
srate = 44100.0;
break;
case '2':
srate = 22050.0;
break;
case '1':
srate = 11025.0;
break;
case '8':
srate = 8000.0;
break;
default:
printf("flag %s ignored\n", argv[i]);
break;
}
} else if (*flag == 'c') {
channels = atoi(flag+1);
if (channels < 1 || channels > 16) {
channels = 1;
printf("flag %s ignored, using 1 channel\n",
argv[i]);
}
}
}
}
if (!tofile) {
printf("2 files not found, use -? for help\n");
return 1;
}
from_snd.device = SND_DEVICE_FILE;
from_snd.write_flag = SND_READ;
from_snd.u.file.byte_offset = 0;
from_snd.format.channels = channels;
from_snd.format.mode = mode;
from_snd.format.bits = bits;
from_snd.format.srate = srate;
if (strcmp(fromfile, "ada") == 0) {
from_snd.device = SND_DEVICE_AUDIO;
from_snd.u.audio.protocol = SND_COMPUTEAHEAD;
from_snd.u.audio.latency = 1.0;
from_snd.u.audio.granularity = 0.1;
strcpy(from_snd.u.file.filename, "");
} else {
strcpy(from_snd.u.file.filename, fromfile);
}
if (snd_open(&from_snd, &flags) != SND_SUCCESS) {
printf("error opening %s for input\n", fromfile);
exit(1);
}
printf("Opened %s for input: ", from_snd.u.file.filename);
snd_print_info(&from_snd);
to_snd.device = SND_DEVICE_FILE;
to_snd.write_flag = SND_WRITE;
to_snd.format.channels = channels;
to_snd.format.mode = mode;
to_snd.format.bits = bits;
to_snd.format.srate = from_snd.format.srate;
to_snd.u.file.header = format; /* header format, e.g. AIFF */
if (to_snd.u.file.header == SND_HEAD_WAVE && to_snd.format.mode == SND_MODE_PCM &&
to_snd.format.bits == 8) to_snd.format.mode = SND_MODE_UPCM;
if (strcmp(tofile, "ada") == 0) {
to_snd.device = SND_DEVICE_AUDIO;
to_snd.u.audio.protocol = SND_COMPUTEAHEAD;
to_snd.u.audio.latency = 0.0;
to_snd.u.audio.granularity = 0.1;
strcpy(to_snd.u.audio.interfacename, "");
strcpy(to_snd.u.audio.devicename, "");
} else {
strcpy(to_snd.u.file.filename, tofile);
}
if (snd_open(&to_snd, &flags) != SND_SUCCESS) {
printf("error opening %s for output\n", tofile);
exit(1);
}
printf("Opened %s for output: ", to_snd.u.file.filename);
snd_print_info(&to_snd);
/* figure out how much to convert on each iteration */
/* first, compute how many frames could fit in each buffer */
from_len = MAXLEN / snd_bytes_per_frame(&from_snd);
to_len = MAXLEN / snd_bytes_per_frame(&to_snd);
/* then compute the minimum of the two frame counts */
frames = min(from_len, to_len);
while (1) {
/* then convert back from frame counts to byte counts: */
while ((frames_from_len = snd_poll(&from_snd)) <=0);
frames_from_len = min(frames_from_len, frames);
while ((frames_to_len = snd_poll(&to_snd)) <= 0);
frames_to_len = min(frames_to_len, frames);
len = min(frames_to_len, frames_from_len);
len = snd_read(&from_snd, from_buf, len);
if (((from_snd.device == SND_DEVICE_AUDIO) &&
(count > from_snd.format.srate * 10)) || (!len)) {
break;
}
for (i = 0; i < len * snd_bytes_per_frame(&from_snd); i++) {
checksum += from_buf[i];
count++;
}
len = snd_convert(&to_snd, to_buf, &from_snd, from_buf, len);
len = snd_write(&to_snd, to_buf, len);
printf("#");
fflush(stdout);
}
snd_close(&from_snd);
if (to_snd.device == SND_DEVICE_AUDIO) {
while (snd_flush(&to_snd) != SND_SUCCESS) ;
}
snd_close(&to_snd);
printf("Read %ld frames, checksum = %ld\n", count, checksum);
exit(0);
}
