static
void pc_speaker_play (pc_speaker_t *spk, uint16_t *buf, unsigned cnt)
{
if (spk->lowpass_freq > 0) {
snd_iir2_filter (&spk->iir, buf, buf, cnt, 1, 1);
}
snd_write (spk->drv, buf, cnt);
}
/*
* 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;
}
}
}
bool ExportPCM(AudacityProject *project,
wxString format, bool stereo, wxString fName,
bool selectionOnly, double t0, double t1)
{
wxMessageBox("In process of being rewritten, sorry...");
#if 0
double rate = project->GetRate();
wxWindow *parent = project;
TrackList *tracks = project->GetTracks();
int header = SND_HEAD_NONE;
#ifdef __WXMAC__
bool trackMarkers = false;
#endif
if (format == "WAV")
header = SND_HEAD_WAVE;
else if (format == "AIFF")
header = SND_HEAD_AIFF;
else if (format == "IRCAM")
header = SND_HEAD_IRCAM;
else if (format == "AU")
header = SND_HEAD_NEXT;
#ifdef __WXMAC__
else if (format == "AIFF with track markers") {
header = SND_HEAD_AIFF;
trackMarkers = true;
}
#endif
// Use snd library to export file
snd_node sndfile;
snd_node sndbuffer;
sndfile.device = SND_DEVICE_FILE;
sndfile.write_flag = SND_WRITE;
strcpy(sndfile.u.file.filename, (const char *) fName);
sndfile.u.file.file = 0;
sndfile.u.file.header = header;
sndfile.u.file.byte_offset = 0;
sndfile.u.file.end_offset = 0;
sndfile.u.file.swap = 0;
sndfile.format.channels = stereo ? 2 : 1;
sndfile.format.mode = SND_MODE_PCM; // SND_MODE_FLOAT
sndfile.format.bits = 16;
sndfile.format.srate = int (rate + 0.5);
int err;
long flags = 0;
err = snd_open(&sndfile, &flags);
if (err) {
wxMessageBox("Could not write to file.");
return false;
}
sndbuffer.device = SND_DEVICE_MEM;
sndbuffer.write_flag = SND_READ;
sndbuffer.u.mem.buffer_max = 0;
sndbuffer.u.mem.buffer = 0;
sndbuffer.u.mem.buffer_len = 0;
sndbuffer.u.mem.buffer_pos = 0;
sndbuffer.format.channels = stereo ? 2 : 1;
sndbuffer.format.mode = SND_MODE_PCM; // SND_MODE_FLOAT
sndbuffer.format.bits = 16;
sndbuffer.format.srate = int (rate + 0.5);
double timeStep = 10.0; // write in blocks of 10 secs
wxProgressDialog *progress = NULL;
wxYield();
wxStartTimer();
wxBusyCursor busy;
bool cancelling = false;
double t = t0;
while (t < t1 && !cancelling) {
double deltat = timeStep;
if (t + deltat > t1)
deltat = t1 - t;
sampleCount numSamples = int (deltat * rate + 0.5);
Mixer *mixer = new Mixer(stereo ? 2 : 1, numSamples, true, rate);
wxASSERT(mixer);
mixer->Clear();
char *buffer = new char[numSamples * 2 * sndbuffer.format.channels];
wxASSERT(buffer);
TrackListIterator iter(tracks);
VTrack *tr = iter.First();
while (tr) {
if (tr->GetKind() == VTrack::Wave) {
if (tr->selected || !selectionOnly) {
if (tr->channel == VTrack::MonoChannel)
mixer->MixMono((WaveTrack *) tr, t, t + deltat);
if (tr->channel == VTrack::LeftChannel)
mixer->MixLeft((WaveTrack *) tr, t, t + deltat);
if (tr->channel == VTrack::RightChannel)
mixer->MixRight((WaveTrack *) tr, t, t + deltat);
}
}
tr = iter.Next();
}
sampleType *mixed = mixer->GetBuffer();
long b2 = snd_convert(&sndfile, buffer, // to
&sndbuffer, mixed, numSamples); // from
snd_write(&sndfile, buffer, b2);
t += deltat;
if (!progress && wxGetElapsedTime(false) > 500) {
wxString message;
if (selectionOnly)
message =
wxString::
Format("Exporting the selected audio as a %s file",
(const char *) format);
else
message =
wxString::
Format("Exporting the entire project as a %s file",
(const char *) format);
progress =
new wxProgressDialog("Export",
message,
1000,
parent,
wxPD_CAN_ABORT |
wxPD_REMAINING_TIME | wxPD_AUTO_HIDE);
}
if (progress) {
cancelling =
!progress->Update(int (((t - t0) * 1000) / (t1 - t0) + 0.5));
}
delete mixer;
delete[]buffer;
}
snd_close(&sndfile);
#ifdef __WXMAC__
FSSpec spec;
wxMacFilename2FSSpec(fName, &spec);
if (trackMarkers) {
// Export the label track as "CD Spin Doctor" files
LabelTrack *labels = NULL;
TrackListIterator iter(tracks);
VTrack *t = iter.First();
while (t && !labels) {
if (t->GetKind() == VTrack::Label)
labels = (LabelTrack *) t;
t = iter.Next();
}
if (labels) {
FSpCreateResFile(&spec, 'AIFF', AUDACITY_CREATOR, 0);
int resFile = FSpOpenResFile(&spec, fsWrPerm);
if (resFile == -1) {
int x = ResError();
}
if (resFile != -1) {
UseResFile(resFile);
int numLabels = labels->mLabels.Count();
for (int i = 0; i < numLabels; i++) {
int startBlock = (int) (labels->mLabels[i]->t * 75);
int lenBlock;
if (i < numLabels - 1)
lenBlock =
(int) ((labels->mLabels[i + 1]->t -
labels->mLabels[i]->t) * 75);
else
lenBlock =
(int) ((tracks->GetMaxLen() -
labels->mLabels[i]->t) * 75);
int startSample = startBlock * 1176 + 54;
int lenSample = lenBlock * 1176 + 54;
Handle theHandle = NewHandle(50);
HLock(theHandle);
char *data = (char *) (*theHandle);
*(int *) &data[0] = startSample;
*(int *) &data[4] = lenSample;
*(int *) &data[8] = startBlock;
*(int *) &data[12] = lenBlock;
*(short *) &data[16] = i + 1;
wxString title = labels->mLabels[i]->title;
if (title.Length() > 31)
title = title.Left(31);
data[18] = title.Length();
strcpy(&data[19], (const char *) title);
HUnlock(theHandle);
AddResource(theHandle, 'SdCv', 128 + i, "\p");
}
CloseResFile(resFile);
wxMessageBox("Saved track information with file.");
}
}
}
FInfo finfo;
if (FSpGetFInfo(&spec, &finfo) == noErr) {
switch (header) {
case SND_HEAD_AIFF:
finfo.fdType = 'AIFF';
break;
case SND_HEAD_IRCAM:
finfo.fdType = 'IRCA';
break;
case SND_HEAD_NEXT:
finfo.fdType = 'AU ';
break;
case SND_HEAD_WAVE:
finfo.fdType = 'WAVE';
break;
}
finfo.fdCreator = AUDACITY_CREATOR;
FSpSetFInfo(&spec, &finfo);
}
#endif
if (progress)
delete progress;
return true;
#endif
return false;
}
sample_type sound_save_array(LVAL sa, long n, snd_type snd,
char *buf, long *ntotal, snd_type player)
{
long i, chans;
long buflen;
sound_state_type state;
double start_time = HUGE_VAL;
float *float_bufp;
LVAL sa_copy;
long debug_unit; /* print messages at intervals of this many samples */
long debug_count; /* next point at which to print a message */
sample_type max_sample = 0.0F;
cvtfn_type cvtfn;
*ntotal = 0;
/* THE ALGORITHM: first merge floating point samples from N channels
* into consecutive multi-channel frames in buf. Then, treat buf
* as just one channel and use one of the cvt_to_* functions to
* convert the data IN PLACE in the buffer (this is ok because the
* converted data will never take more space than the original 32-bit
* floats, so the converted data will not overwrite any floats before
* the floats are converted
*/
/* if snd_expr was simply a symbol, then sa now points to
a shared sound_node. If we read samples from it, then
the sounds bound to the symbol will be destroyed, so
copy it first. If snd_expr was a real expression that
computed a new value, then the next garbage collection
will reclaim the sound array. See also sound_save_sound()
*/
chans = getsize(sa);
if (chans > MAX_SND_CHANNELS) {
xlerror("sound_save: too many channels", sa);
free(buf);
snd_close(snd);
}
xlprot1(sa);
sa_copy = newvector(chans);
xlprot1(sa_copy);
/* Why do we copy the array into an xlisp array instead of just
* the state[i] array? Because some of these sounds may reference
* the lisp heap. We must put the sounds in an xlisp array so that
* the gc will find and mark them. xlprot1(sa_copy) makes the array
* visible to gc.
*/
for (i = 0; i < chans; i++) {
sound_type s = getsound(getelement(sa, i));
setelement(sa_copy, i, cvsound(sound_copy(s)));
}
sa = sa_copy; /* destroy original reference to allow GC */
state = (sound_state_type) malloc(sizeof(sound_state_node) * chans);
for (i = 0; i < chans; i++) {
state[i].sound = getsound(getelement(sa, i));
state[i].scale = state[i].sound->scale;
D nyquist_printf("save scale factor %d = %g\n", (int)i, state[i].scale);
state[i].terminated = false;
state[i].cnt = 0; /* force a fetch */
start_time = min(start_time, state[i].sound->t0);
}
for (i = 0; i < chans; i++) {
if (state[i].sound->t0 > start_time)
sound_prepend_zeros(state[i].sound, start_time);
}
/* for debugging */
/* printing_this_sound = s;*/
cvtfn = find_cvt_to_fn(snd, buf);
#ifdef MACINTOSH
if (player) {
gprintf(TRANS, "Playing audio: Click and hold mouse button to stop playback.\n");
}
#endif
debug_unit = debug_count = (long) max(snd->format.srate, 10000.0);
while (n > 0) {
/* keep the following information for each sound:
has it terminated?
pointer to samples
number of samples remaining in block
scan to find the minimum remaining samples and
output that many in an inner loop. Stop outer
loop if all sounds have terminated
*/
int terminated = true;
int togo = n;
int j;
float peak;
oscheck();
for (i = 0; i < chans; i++) {
if (state[i].cnt == 0) {
if (sndwrite_trace) {
nyquist_printf("CALLING SOUND_GET_NEXT "
"ON CHANNEL %d (%p)\n",
(int)i, state[i].sound);
sound_print_tree(state[i].sound);
}
state[i].ptr = sound_get_next(state[i].sound,
&(state[i].cnt))->samples;
if (sndwrite_trace) {
nyquist_printf("RETURNED FROM CALL TO SOUND_GET_NEXT "
"ON CHANNEL %d\n", (int)i);
}
if (state[i].ptr == zero_block->samples) {
state[i].terminated = true;
}
}
if (!state[i].terminated) terminated = false;
togo = min(togo, state[i].cnt);
}
if (terminated) break;
float_bufp = (float *) buf;
for (j = 0; j < togo; j++) {
for (i = 0; i < chans; i++) {
double s = *(state[i].ptr++) * state[i].scale;
*float_bufp++ = (float) s;
}
}
// we're treating sound as mono for the conversion, so multiply
// togo by chans to get proper number of samples, and divide by
// chans to convert back to frame count required by snd_write
buflen = (*cvtfn)((void *) buf, (void *) buf, togo * chans, 1.0F,
&peak) / chans;
if (peak > max_sample) max_sample = peak;
#ifdef MACINTOSH
if (Button()) {
if (player) {
snd_reset(player);
}
gprintf(TRANS, "\n\nStopping playback...\n\n\n");
break;
}
#endif
if (snd->u.file.file != -1) snd_write(snd, (void *) buf, buflen);
if (player) write_to_audio(player, (void *) buf, buflen);
n -= togo;
for (i = 0; i < chans; i++) {
state[i].cnt -= togo;
}
*ntotal += togo;
if (*ntotal > debug_count) {
gprintf(TRANS, " %d ", *ntotal);
fflush(stdout);
debug_count += debug_unit;
}
}
gprintf(TRANS, "total samples: %d x %d channels\n",
*ntotal, chans);
/* references to sounds are shared by sa_copy and state[].
* here, we dispose of state[], allowing GC to do the
* sound_unref call that frees the sounds. (Freeing them now
* would be a bug.)
*/
free(state);
xlpop();
return max_sample;
}
sample_type sound_save_sound(LVAL s_as_lval, long n, snd_type snd,
char *buf, long *ntotal, snd_type player)
{
long blocklen;
long buflen;
sound_type s;
long debug_unit; /* print messages at intervals of this many samples */
long debug_count; /* next point at which to print a message */
sample_type max_sample = 0.0F;
cvtfn_type cvtfn;
*ntotal = 0;
/* if snd_expr was simply a symbol, then s now points to
a shared sound_node. If we read samples from it, then
the sound bound to the symbol will be destroyed, so
copy it first. If snd_expr was a real expression that
computed a new value, then the next garbage collection
will reclaim the sound_node. We need to make the new
sound reachable by the garbage collector to that any
lisp data reachable from the sound do not get collected.
To make the sound reachable, we need to allocate a node,
and the GC might run, so we need to protect the OLD s
but then make it unreachable.
We will let the GC collect the sound in the end.
*/
xlprot1(s_as_lval);
s = sound_copy(getsound(s_as_lval));
s_as_lval = cvsound(s); /* destroys only ref. to original */
/* for debugging */
/* printing_this_sound = s;*/
debug_unit = debug_count = (long) max(snd->format.srate, 10000.0);
cvtfn = find_cvt_to_fn(snd, buf);
#ifdef MACINTOSH
if (player) {
gprintf(TRANS, "Playing audio: Click and hold mouse button to stop playback.\n");
}
#endif
while (n > 0) {
long togo;
float peak;
sample_block_type sampblock = sound_get_next(s, &blocklen);
oscheck();
#ifdef SNAPSHOTS
stdputstr(".");
if (sound_created_flag) {
stdputstr("SNAPSHOT: ");
sound_print_tree(printing_this_sound);
sound_created_flag = false;
}
fflush(stdout);
#endif
if (sampblock == zero_block || blocklen == 0) {
break;
}
togo = min(blocklen, n);
buflen = (*cvtfn)((void *) buf, (void *) sampblock->samples,
togo, s->scale, &peak);
if (peak > max_sample) max_sample = peak;
#ifdef MACINTOSH
if (Button()) {
if (player) {
snd_reset(player);
}
gprintf(TRANS, "\n\nStopping playback...\n\n\n");
break;
}
#endif
if (snd->u.file.file != -1) snd_write(snd, (void *) buf, buflen);
if (player) write_to_audio(player, (void *) buf, buflen);
n -= togo;
*ntotal += togo;
if (*ntotal > debug_count) {
gprintf(TRANS, " %d ", *ntotal);
fflush(stdout);
debug_count += debug_unit;
}
}
gprintf(TRANS, "\ntotal samples: %d\n", *ntotal);
xlpop();
return max_sample;
}
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);
}
