int SNDWAV_PrepareWAVParams(WAVContainer_t *wav)
{
assert(wav);
//硬件参数赋值
uint16_t channels = DEFAULT_CHANNELS;
uint16_t sample_rate = DEFAULT_SAMPLE_RATE;
uint16_t sample_length = DEFAULT_SAMPLE_LENGTH;
uint32_t duration_time = DEFAULT_DURATION_TIME;
wav->header.magic = WAV_RIFF;
wav->header.type = WAV_WAVE;
wav->format.magic = WAV_FMT;
wav->format.fmt_size = LE_INT(16);
wav->format.format = LE_SHORT(WAV_FMT_PCM);
wav->chunk.type = WAV_DATA;
//自定义
wav->format.channels = LE_SHORT(channels);
wav->format.sample_rate = LE_INT(sample_rate);
wav->format.sample_length = LE_SHORT(sample_length);
wav->format.blocks_align = LE_SHORT(channels * sample_length / 8);
wav->format.bytes_p_second = LE_INT((uint16_t)(wav->format.blocks_align) * sample_rate);
wav->chunk.length = LE_INT(duration_time * (uint32_t)(wav->format.bytes_p_second));
wav->header.length = LE_INT((uint32_t)(wav->chunk.length) + sizeof(wav->chunk) + sizeof(wav->format) + sizeof(wav->header) - 8);
return 0;
}
static int READW(uint16_t *vp, struct timidity_file *tf)
{
if (tf_read(vp, 2, 1, tf) != 1)
return -1;
*vp = LE_SHORT(*vp);
return 1;
}
int SNDWAV_P_GetFormat(WAVContainer_t *wav, snd_pcm_format_t *snd_format)
{
if (LE_SHORT(wav->format.format) != WAV_FMT_PCM)
return -1;
switch (LE_SHORT(wav->format.sample_length)) {
case 16:
*snd_format = SND_PCM_FORMAT_S16_LE;
break;
case 8:
*snd_format = SND_PCM_FORMAT_U8;
break;
default:
*snd_format = SND_PCM_FORMAT_UNKNOWN;
break;
}
return 0;
}
int CWAVInputSource::AnalyzeSource()
{
unsigned char *p = FULL_HEADER, *priff = NULL;
// seek to the beginning (just in case)
m_spIO->Seek(0, FILE_BEGIN);
// get the file size
m_nFileBytes = m_spIO->GetSize();
// get the RIFF header
RETURN_ON_ERROR(ReadSafe(m_spIO, p, sizeof(RIFF_HEADER)))
// make sure the RIFF header is valid
if (!(p[0] == 'R' && p[1] == 'I' && p[2] == 'F' && p[3] == 'F'))
return ERROR_INVALID_INPUT_FILE;
p += sizeof(RIFF_HEADER);
// read the data type header
RETURN_ON_ERROR(ReadSafe(m_spIO, p, sizeof(DATA_TYPE_ID_HEADER)))
// make sure it's the right data type
if (!(p[0] == 'W' && p[1] == 'A' && p[2] == 'V' && p[3] == 'E'))
return ERROR_INVALID_INPUT_FILE;
p += sizeof(DATA_TYPE_ID_HEADER);
// find the 'fmt ' chunk
RETURN_ON_ERROR(ReadSafe(m_spIO, p, sizeof(RIFF_CHUNK_HEADER)))
while (!(p[0] == 'f' && p[1] == 'm' && p[2] == 't' && p[3] == ' '))
{
p += sizeof(RIFF_CHUNK_HEADER);
// move the file pointer to the end of this chunk
RETURN_ON_ERROR(ReadSafe(m_spIO, p, LE_LONG(p+4)))
p += LE_LONG(p+4);
// check again for the data chunk
RETURN_ON_ERROR(ReadSafe(m_spIO, p, sizeof(RIFF_CHUNK_HEADER)))
}
priff = p+4;
p += sizeof(RIFF_CHUNK_HEADER);
// read the format info
RETURN_ON_ERROR(ReadSafe(m_spIO, p, sizeof(WAV_FORMAT_HEADER)))
// error check the header to see if we support it
if (LE_SHORT(p) != 1)
return ERROR_INVALID_INPUT_FILE;
// copy the format information to the WAVEFORMATEX passed in
FillWaveFormatEx(&m_wfeSource, LE_LONG(p+4), LE_SHORT(p+14), LE_SHORT(p+2));
p += sizeof(WAV_FORMAT_HEADER);
// skip over any extra data in the header
int nWAVFormatHeaderExtra = LE_LONG(priff) - sizeof(WAV_FORMAT_HEADER);
if (nWAVFormatHeaderExtra < 0)
return ERROR_INVALID_INPUT_FILE;
else {
RETURN_ON_ERROR(ReadSafe(m_spIO, p, nWAVFormatHeaderExtra))
p += nWAVFormatHeaderExtra;
}
// find the data chunk
RETURN_ON_ERROR(ReadSafe(m_spIO, p, sizeof(RIFF_CHUNK_HEADER)))
while (!(p[0] == 'd' && p[1] == 'a' && p[2] == 't' && p[3] == 'a'))
{
p += sizeof(RIFF_CHUNK_HEADER);
// move the file pointer to the end of this chunk
RETURN_ON_ERROR(ReadSafe(m_spIO, p, LE_LONG(p+4)))
p += LE_LONG(p+4);
// check again for the data chunk
RETURN_ON_ERROR(ReadSafe(m_spIO, p, sizeof(RIFF_CHUNK_HEADER)))
}
// we're at the data block
m_nDataBytes = LE_LONG(p+4);
if (m_nDataBytes < 0)
m_nDataBytes = m_nFileBytes - m_nHeaderBytes;
p += sizeof(RIFF_CHUNK_HEADER);
m_nHeaderBytes = p - FULL_HEADER;
// make sure the data bytes is a whole number of blocks
if ((m_nDataBytes % m_wfeSource.nBlockAlign) != 0)
return ERROR_INVALID_INPUT_FILE;
// calculate the terminating byts
m_nTerminatingBytes = 0;
// we made it this far, everything must be cool
return ERROR_SUCCESS;
}
// test, if it's a .WAV file, > 0 if ok (and set the speed, stereo etc.)
// == 0 if not
// Value returned is bytes to be discarded.
static ssize_t test_wavefile(int fd, u_char *_buffer, size_t size)
{
WaveHeader *h = (WaveHeader *)_buffer;
u_char *buffer = NULL;
size_t blimit = 0;
WaveFmtBody *f;
WaveChunkHeader *c;
u_int type, len;
if (size < sizeof(WaveHeader))
return -1;
if (h->magic != WAV_RIFF || h->type != WAV_WAVE)
return -1;
if (size > sizeof(WaveHeader)) {
check_wavefile_space(buffer, size - sizeof(WaveHeader), blimit);
memcpy(buffer, _buffer + sizeof(WaveHeader), size - sizeof(WaveHeader));
}
size -= sizeof(WaveHeader);
while (1) {
check_wavefile_space(buffer, sizeof(WaveChunkHeader), blimit);
test_wavefile_read(fd, buffer, &size, sizeof(WaveChunkHeader), __LINE__);
c = (WaveChunkHeader*)buffer;
type = c->type;
len = LE_INT(c->length);
len += len % 2;
if (size > sizeof(WaveChunkHeader))
memmove(buffer, buffer + sizeof(WaveChunkHeader), size - sizeof(WaveChunkHeader));
size -= sizeof(WaveChunkHeader);
if (type == WAV_FMT)
break;
check_wavefile_space(buffer, len, blimit);
test_wavefile_read(fd, buffer, &size, len, __LINE__);
if (size > len)
memmove(buffer, buffer + len, size - len);
size -= len;
}
if (len < sizeof(WaveFmtBody)) {
error(_("unknown length of 'fmt ' chunk (read %u, should be %u at least)"),
len, (u_int)sizeof(WaveFmtBody));
exit(EXIT_FAILURE);
}
check_wavefile_space(buffer, len, blimit);
test_wavefile_read(fd, buffer, &size, len, __LINE__);
f = (WaveFmtBody*) buffer;
if (LE_SHORT(f->format) != WAV_PCM_CODE) {
error(_("can't play not PCM-coded WAVE-files"));
exit(EXIT_FAILURE);
}
if (LE_SHORT(f->modus) < 1) {
error(_("can't play WAVE-files with %d tracks"), LE_SHORT(f->modus));
exit(EXIT_FAILURE);
}
hwparams.channels = LE_SHORT(f->modus);
switch (LE_SHORT(f->bit_p_spl)) {
case 8:
if (hwparams.format != DEFAULT_FORMAT &&
hwparams.format != SND_PCM_FORMAT_U8)
fprintf(stderr, _("Warning: format is changed to U8\n"));
hwparams.format = SND_PCM_FORMAT_U8;
break;
case 16:
if (hwparams.format != DEFAULT_FORMAT &&
hwparams.format != SND_PCM_FORMAT_S16_LE)
fprintf(stderr, _("Warning: format is changed to S16_LE\n"));
hwparams.format = SND_PCM_FORMAT_S16_LE;
break;
case 24:
switch (LE_SHORT(f->byte_p_spl) / hwparams.channels) {
case 3:
if (hwparams.format != DEFAULT_FORMAT &&
hwparams.format != SND_PCM_FORMAT_S24_3LE)
fprintf(stderr, _("Warning: format is changed to S24_3LE\n"));
hwparams.format = SND_PCM_FORMAT_S24_3LE;
break;
case 4:
if (hwparams.format != DEFAULT_FORMAT &&
hwparams.format != SND_PCM_FORMAT_S24_LE)
fprintf(stderr, _("Warning: format is changed to S24_LE\n"));
hwparams.format = SND_PCM_FORMAT_S24_LE;
break;
default:
error(_(" can't play WAVE-files with sample %d bits in %d bytes wide (%d channels)"),
LE_SHORT(f->bit_p_spl), LE_SHORT(f->byte_p_spl), hwparams.channels);
exit(EXIT_FAILURE);
}
break;
case 32:
hwparams.format = SND_PCM_FORMAT_S32_LE;
break;
default:
error(_(" can't play WAVE-files with sample %d bits wide"),
LE_SHORT(f->bit_p_spl));
exit(EXIT_FAILURE);
}
hwparams.rate = LE_INT(f->sample_fq);
if (size > len)
memmove(buffer, buffer + len, size - len);
size -= len;
while (1) {
u_int type, len;
check_wavefile_space(buffer, sizeof(WaveChunkHeader), blimit);
test_wavefile_read(fd, buffer, &size, sizeof(WaveChunkHeader), __LINE__);
c = (WaveChunkHeader*)buffer;
type = c->type;
len = LE_INT(c->length);
if (size > sizeof(WaveChunkHeader))
memmove(buffer, buffer + sizeof(WaveChunkHeader), size - sizeof(WaveChunkHeader));
size -= sizeof(WaveChunkHeader);
if (type == WAV_DATA) {
if (len < pbrec_count && len < 0x7ffffffe)
pbrec_count = len;
if (size > 0)
memcpy(_buffer, buffer, size);
free(buffer);
return size;
}
len += len % 2;
check_wavefile_space(buffer, len, blimit);
test_wavefile_read(fd, buffer, &size, len, __LINE__);
if (size > len)
memmove(buffer, buffer + len, size - len);
size -= len;
}
// shouldn't be reached
return -1;
}
/* write a WAVE-header */
static void begin_wave(int fd, size_t cnt)
{
WaveHeader h;
WaveFmtBody f;
WaveChunkHeader cf, cd;
int bits;
u_int tmp;
u_short tmp2;
/* WAVE cannot handle greater than 32bit (signed?) int */
if (cnt == (size_t)-2)
cnt = 0x7fffff00;
bits = 8;
switch ((unsigned long) hwparams.format) {
case SND_PCM_FORMAT_U8:
bits = 8;
break;
case SND_PCM_FORMAT_S16_LE:
bits = 16;
break;
case SND_PCM_FORMAT_S32_LE:
case SND_PCM_FORMAT_FLOAT_LE:
bits = 32;
break;
case SND_PCM_FORMAT_S24_LE:
case SND_PCM_FORMAT_S24_3LE:
bits = 24;
break;
default:
error(_("Wave doesn't support %s format..."), snd_pcm_format_name(hwparams.format));
exit(EXIT_FAILURE);
}
h.magic = WAV_RIFF;
tmp = cnt + sizeof(WaveHeader) + sizeof(WaveChunkHeader) + sizeof(WaveFmtBody) + sizeof(WaveChunkHeader) - 8;
h.length = LE_INT(tmp);
h.type = WAV_WAVE;
cf.type = WAV_FMT;
cf.length = LE_INT(16);
if (hwparams.format == SND_PCM_FORMAT_FLOAT_LE)
f.format = LE_SHORT(WAV_FMT_IEEE_FLOAT);
else
f.format = LE_SHORT(WAV_FMT_PCM);
f.channels = LE_SHORT(hwparams.channels);
f.sample_fq = LE_INT(hwparams.rate);
#if 0
tmp2 = (samplesize == 8) ? 1 : 2;
f.byte_p_spl = LE_SHORT(tmp2);
tmp = dsp_speed * hwparams.channels * (u_int) tmp2;
#else
tmp2 = hwparams.channels * snd_pcm_format_physical_width(hwparams.format) / 8;
f.byte_p_spl = LE_SHORT(tmp2);
tmp = (u_int) tmp2 * hwparams.rate;
#endif
f.byte_p_sec = LE_INT(tmp);
f.bit_p_spl = LE_SHORT(bits);
cd.type = WAV_DATA;
cd.length = LE_INT(cnt);
if (write(fd, &h, sizeof(WaveHeader)) != sizeof(WaveHeader) ||
write(fd, &cf, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader) ||
write(fd, &f, sizeof(WaveFmtBody)) != sizeof(WaveFmtBody) ||
write(fd, &cd, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader)) {
error(_("write error"));
exit(EXIT_FAILURE);
}
}
int ELD::update_eld(const char *buf, int size)
{
int mnl;
int i;
m_e.eld_ver = GRAB_BITS(buf, 0, 3, 5);
if (m_e.eld_ver != ELD_VER_CEA_861D &&
m_e.eld_ver != ELD_VER_PARTIAL)
{
VBAUDIO(QString("Unknown ELD version %1").arg(m_e.eld_ver));
goto out_fail;
}
m_e.eld_size = size;
m_e.baseline_len = GRAB_BITS(buf, 2, 0, 8);
mnl = GRAB_BITS(buf, 4, 0, 5);
m_e.cea_edid_ver = GRAB_BITS(buf, 4, 5, 3);
m_e.support_hdcp = GRAB_BITS(buf, 5, 0, 1);
m_e.support_ai = GRAB_BITS(buf, 5, 1, 1);
m_e.conn_type = GRAB_BITS(buf, 5, 2, 2);
m_e.sad_count = GRAB_BITS(buf, 5, 4, 4);
m_e.aud_synch_delay = GRAB_BITS(buf, 6, 0, 8) * 2;
m_e.spk_alloc = GRAB_BITS(buf, 7, 0, 7);
m_e.port_id = LE_INT64(buf + 8);
/* not specified, but the spec's tendency is little endian */
m_e.manufacture_id = LE_SHORT(buf + 16);
m_e.product_id = LE_SHORT(buf + 18);
if (mnl > ELD_MAX_MNL)
{
VBAUDIO(QString("MNL is reserved value %1").arg(mnl));
goto out_fail;
}
else if (ELD_FIXED_BYTES + mnl > size)
{
VBAUDIO(QString("out of range MNL %1").arg(mnl));
goto out_fail;
}
else
{
strncpy(m_e.monitor_name, (char *)buf + ELD_FIXED_BYTES, mnl + 1);
m_e.monitor_name[mnl] = '\0';
}
for (i = 0; i < m_e.sad_count; i++)
{
if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size)
{
VBAUDIO(QString("out of range SAD %1").arg(i));
goto out_fail;
}
update_sad(i, buf + ELD_FIXED_BYTES + mnl + 3 * i);
}
/*
* Assume the highest speakers configuration
*/
if (!m_e.spk_alloc)
m_e.spk_alloc = 0xffff;
m_e.eld_valid = true;
return 0;
out_fail:
m_e.eld_valid = false;
return -1;
}
/*
If panning or note_to_use != -1, it will be used for all samples,
instead of the sample-specific values in the instrument file.
For note_to_use, any value <0 or >127 will be forced to 0.
For other parameters, 1 means yes, 0 means no, other values are
undefined.
TODO: do reverse loops right */
static InstrumentLayer *load_instrument(char *name, int font_type, int percussion,
int panning, int amp, int cfg_tuning, int note_to_use,
int strip_loop, int strip_envelope,
int strip_tail, int bank, int gm_num, int sf_ix)
{
InstrumentLayer *lp, *lastlp, *headlp;
Instrument *ip;
FILE *fp;
uint8 tmp[1024];
int i,j,noluck=0;
#ifdef PATCH_EXT_LIST
static char *patch_ext[] = PATCH_EXT_LIST;
#endif
int sf2flag = 0;
int right_samples = 0;
int stereo_channels = 1, stereo_layer;
int vlayer_list[19][4], vlayer, vlayer_count;
if (!name) return 0;
/* Open patch file */
if ((fp=open_file(name, 1, OF_NORMAL)) == NULL)
{
noluck=1;
#ifdef PATCH_EXT_LIST
/* Try with various extensions */
for (i=0; patch_ext[i]; i++)
{
if (strlen(name)+strlen(patch_ext[i])<1024)
{
char path[1024];
strcpy(path, name);
strcat(path, patch_ext[i]);
if ((fp=open_file(path, 1, OF_NORMAL)) != NULL)
{
noluck=0;
break;
}
}
}
#endif
}
if (noluck)
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Instrument `%s' can't be found.", name);
return 0;
}
/*ctl->cmsg(CMSG_INFO, VERB_NOISY, "Loading instrument %s", current_filename);*/
/* Read some headers and do cursory sanity checks. There are loads
of magic offsets. This could be rewritten... */
if ((239 != fread(tmp, 1, 239, fp)) ||
(memcmp(tmp, "GF1PATCH110\0ID#000002", 22) &&
memcmp(tmp, "GF1PATCH100\0ID#000002", 22))) /* don't know what the
differences are */
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s: not an instrument", name);
return 0;
}
/* patch layout:
* bytes: info: starts at offset:
* 22 id (see above) 0
* 60 copyright 22
* 1 instruments 82
* 1 voices 83
* 1 channels 84
* 2 number of waveforms 85
* 2 master volume 87
* 4 datasize 89
* 36 reserved, but now: 93
* 7 "SF2EXT\0" id 93
* 1 right samples 100
* 28 reserved 101
* 2 instrument number 129
* 16 instrument name 131
* 4 instrument size 147
* 1 number of layers 151
* 40 reserved 152
* 1 layer duplicate 192
* 1 layer number 193
* 4 layer size 194
* 1 number of samples 198
* 40 reserved 199
* 239
* THEN, for each sample, see below
*/
if (!memcmp(tmp + 93, "SF2EXT", 6))
{
sf2flag = 1;
vlayer_count = tmp[152];
}
if (tmp[82] != 1 && tmp[82] != 0) /* instruments. To some patch makers,
0 means 1 */
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Can't handle patches with %d instruments", tmp[82]);
return 0;
}
if (tmp[151] != 1 && tmp[151] != 0) /* layers. What's a layer? */
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Can't handle instruments with %d layers", tmp[151]);
return 0;
}
if (sf2flag && vlayer_count > 0) {
for (i = 0; i < 9; i++)
for (j = 0; j < 4; j++)
vlayer_list[i][j] = tmp[153+i*4+j];
for (i = 9; i < 19; i++)
for (j = 0; j < 4; j++)
vlayer_list[i][j] = tmp[199+(i-9)*4+j];
}
else {
for (i = 0; i < 19; i++)
for (j = 0; j < 4; j++)
vlayer_list[i][j] = 0;
vlayer_list[0][0] = 0;
vlayer_list[0][1] = 127;
vlayer_list[0][2] = tmp[198];
vlayer_list[0][3] = 0;
vlayer_count = 1;
}
lastlp = 0;
for (vlayer = 0; vlayer < vlayer_count; vlayer++) {
lp=(InstrumentLayer *)safe_malloc(sizeof(InstrumentLayer));
lp->size = sizeof(InstrumentLayer);
lp->lo = vlayer_list[vlayer][0];
lp->hi = vlayer_list[vlayer][1];
ip=(Instrument *)safe_malloc(sizeof(Instrument));
lp->size += sizeof(Instrument);
lp->instrument = ip;
lp->next = 0;
if (lastlp) lastlp->next = lp;
else headlp = lp;
lastlp = lp;
if (sf2flag) ip->type = INST_SF2;
else ip->type = INST_GUS;
ip->samples = vlayer_list[vlayer][2];
ip->sample = (Sample *)safe_malloc(sizeof(Sample) * ip->samples);
lp->size += sizeof(Sample) * ip->samples;
ip->left_samples = ip->samples;
ip->left_sample = ip->sample;
right_samples = vlayer_list[vlayer][3];
ip->right_samples = right_samples;
if (right_samples)
{
ip->right_sample = (Sample *)safe_malloc(sizeof(Sample) * right_samples);
lp->size += sizeof(Sample) * right_samples;
stereo_channels = 2;
}
else ip->right_sample = 0;
ip->contents = 0;
ctl->cmsg(CMSG_INFO, VERB_NOISY, "%s%s[%d,%d] %s(%d-%d layer %d of %d)",
(percussion)? " ":"", name,
(percussion)? note_to_use : gm_num, bank,
(right_samples)? "(2) " : "",
lp->lo, lp->hi, vlayer+1, vlayer_count);
for (stereo_layer = 0; stereo_layer < stereo_channels; stereo_layer++)
{
int sample_count;
if (stereo_layer == 0) sample_count = ip->left_samples;
else if (stereo_layer == 1) sample_count = ip->right_samples;
for (i=0; i < sample_count; i++)
{
uint8 fractions;
int32 tmplong;
uint16 tmpshort;
uint16 sample_volume;
uint8 tmpchar;
Sample *sp;
uint8 sf2delay;
#define READ_CHAR(thing) \
if (1 != fread(&tmpchar, 1, 1, fp)) goto fail; \
thing = tmpchar;
#define READ_SHORT(thing) \
if (1 != fread(&tmpshort, 2, 1, fp)) goto fail; \
thing = LE_SHORT(tmpshort);
#define READ_LONG(thing) \
if (1 != fread(&tmplong, 4, 1, fp)) goto fail; \
thing = LE_LONG(tmplong);
/*
* 7 sample name
* 1 fractions
* 4 length
* 4 loop start
* 4 loop end
* 2 sample rate
* 4 low frequency
* 4 high frequency
* 2 finetune
* 1 panning
* 6 envelope rates |
* 6 envelope offsets | 18 bytes
* 3 tremolo sweep, rate, depth |
* 3 vibrato sweep, rate, depth |
* 1 sample mode
* 2 scale frequency
* 2 scale factor
* 2 sample volume (??)
* 34 reserved
* Now: 1 delay
* 33 reserved
*/
skip(fp, 7); /* Skip the wave name */
if (1 != fread(&fractions, 1, 1, fp))
{
fail:
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "Error reading sample %d", i);
if (stereo_layer == 1)
{
for (j=0; j<i; j++)
free(ip->right_sample[j].data);
free(ip->right_sample);
i = ip->left_samples;
}
for (j=0; j<i; j++)
free(ip->left_sample[j].data);
free(ip->left_sample);
free(ip);
free(lp);
return 0;
}
if (stereo_layer == 0) sp=&(ip->left_sample[i]);
else if (stereo_layer == 1) sp=&(ip->right_sample[i]);
READ_LONG(sp->data_length);
READ_LONG(sp->loop_start);
READ_LONG(sp->loop_end);
READ_SHORT(sp->sample_rate);
READ_LONG(sp->low_freq);
READ_LONG(sp->high_freq);
READ_LONG(sp->root_freq);
skip(fp, 2); /* Why have a "root frequency" and then "tuning"?? */
READ_CHAR(tmp[0]);
if (panning==-1)
sp->panning = (tmp[0] * 8 + 4) & 0x7f;
else
sp->panning=(uint8)(panning & 0x7F);
sp->resonance=0;
sp->cutoff_freq=0;
sp->reverberation=0;
sp->chorusdepth=0;
sp->exclusiveClass=0;
sp->keyToModEnvHold=0;
sp->keyToModEnvDecay=0;
sp->keyToVolEnvHold=0;
sp->keyToVolEnvDecay=0;
if (cfg_tuning)
{
double tune_factor = (double)(cfg_tuning)/1200.0;
tune_factor = pow(2.0, tune_factor);
sp->root_freq = (uint32)( tune_factor * (double)sp->root_freq );
}
/* envelope, tremolo, and vibrato */
if (18 != fread(tmp, 1, 18, fp)) goto fail;
if (!tmp[13] || !tmp[14])
{
sp->tremolo_sweep_increment=
sp->tremolo_phase_increment=sp->tremolo_depth=0;
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * no tremolo");
}
else
{
sp->tremolo_sweep_increment=convert_tremolo_sweep(tmp[12]);
sp->tremolo_phase_increment=convert_tremolo_rate(tmp[13]);
sp->tremolo_depth=tmp[14];
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" * tremolo: sweep %d, phase %d, depth %d",
sp->tremolo_sweep_increment, sp->tremolo_phase_increment,
sp->tremolo_depth);
}
if (!tmp[16] || !tmp[17])
{
sp->vibrato_sweep_increment=
sp->vibrato_control_ratio=sp->vibrato_depth=0;
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * no vibrato");
}
else
{
sp->vibrato_control_ratio=convert_vibrato_rate(tmp[16]);
sp->vibrato_sweep_increment=
convert_vibrato_sweep(tmp[15], sp->vibrato_control_ratio);
sp->vibrato_depth=tmp[17];
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" * vibrato: sweep %d, ctl %d, depth %d",
sp->vibrato_sweep_increment, sp->vibrato_control_ratio,
sp->vibrato_depth);
}
READ_CHAR(sp->modes);
READ_SHORT(sp->freq_center);
READ_SHORT(sp->freq_scale);
if (sf2flag)
{
READ_SHORT(sample_volume);
READ_CHAR(sf2delay);
READ_CHAR(sp->exclusiveClass);
skip(fp, 32);
}
else
{
skip(fp, 36);
}
/* Mark this as a fixed-pitch instrument if such a deed is desired. */
if (note_to_use!=-1)
sp->note_to_use=(uint8)(note_to_use);
else
sp->note_to_use=0;
/* seashore.pat in the Midia patch set has no Sustain. I don't
understand why, and fixing it by adding the Sustain flag to
all looped patches probably breaks something else. We do it
anyway. */
if (sp->modes & MODES_LOOPING)
sp->modes |= MODES_SUSTAIN;
/* Strip any loops and envelopes we're permitted to */
if ((strip_loop==1) &&
(sp->modes & (MODES_SUSTAIN | MODES_LOOPING |
MODES_PINGPONG | MODES_REVERSE)))
{
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Removing loop and/or sustain");
sp->modes &=~(MODES_SUSTAIN | MODES_LOOPING |
MODES_PINGPONG | MODES_REVERSE);
}
if (strip_envelope==1)
{
if (sp->modes & MODES_ENVELOPE)
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Removing envelope");
sp->modes &= ~MODES_ENVELOPE;
}
else if (strip_envelope != 0)
{
/* Have to make a guess. */
if (!(sp->modes & (MODES_LOOPING | MODES_PINGPONG | MODES_REVERSE)))
{
/* No loop? Then what's there to sustain? No envelope needed
either... */
sp->modes &= ~(MODES_SUSTAIN|MODES_ENVELOPE);
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - No loop, removing sustain and envelope");
}
else if (!memcmp(tmp, "??????", 6) || tmp[11] >= 100)
{
/* Envelope rates all maxed out? Envelope end at a high "offset"?
That's a weird envelope. Take it out. */
sp->modes &= ~MODES_ENVELOPE;
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - Weirdness, removing envelope");
}
else if (!(sp->modes & MODES_SUSTAIN))
{
/* No sustain? Then no envelope. I don't know if this is
justified, but patches without sustain usually don't need the
envelope either... at least the Gravis ones. They're mostly
drums. I think. */
sp->modes &= ~MODES_ENVELOPE;
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - No sustain, removing envelope");
}
}
sp->attenuation = 0;
for (j=ATTACK; j<DELAY; j++)
{
sp->envelope_rate[j]=
(j<3)? convert_envelope_rate_attack(tmp[j], 11) : convert_envelope_rate(tmp[j]);
sp->envelope_offset[j]=
convert_envelope_offset(tmp[6+j]);
}
if (sf2flag)
{
if (sf2delay > 5) sf2delay = 5;
sp->envelope_rate[DELAY] = (int32)( (sf2delay*play_mode->rate) / 1000 );
}
else
{
sp->envelope_rate[DELAY]=0;
}
sp->envelope_offset[DELAY]=0;
for (j=ATTACK; j<DELAY; j++)
{
sp->modulation_rate[j]=sp->envelope_rate[j];
sp->modulation_offset[j]=sp->envelope_offset[j];
}
sp->modulation_rate[DELAY] = sp->modulation_offset[DELAY] = 0;
sp->modEnvToFilterFc=0;
sp->modEnvToPitch=0;
sp->lfo_sweep_increment = 0;
sp->lfo_phase_increment = 0;
sp->modLfoToFilterFc = 0;
sp->vibrato_delay = 0;
/* Then read the sample data */
if (sp->data_length/2 > MAX_SAMPLE_SIZE)
{
goto fail;
}
sp->data = safe_malloc(sp->data_length + 1);
lp->size += sp->data_length + 1;
if (1 != fread(sp->data, sp->data_length, 1, fp))
goto fail;
if (!(sp->modes & MODES_16BIT)) /* convert to 16-bit data */
{
int32 i=sp->data_length;
uint8 *cp=(uint8 *)(sp->data);
uint16 *tmp,*newdta;
tmp=newdta=safe_malloc(sp->data_length*2 + 2);
while (i--)
*tmp++ = (uint16)(*cp++) << 8;
cp=(uint8 *)(sp->data);
sp->data = (sample_t *)newdta;
free(cp);
sp->data_length *= 2;
sp->loop_start *= 2;
sp->loop_end *= 2;
}
#ifndef LITTLE_ENDIAN
else
/* convert to machine byte order */
{
int32 i=sp->data_length/2;
int16 *tmp=(int16 *)sp->data,s;
while (i--)
{
s=LE_SHORT(*tmp);
*tmp++=s;
}
}
#endif
if (sp->modes & MODES_UNSIGNED) /* convert to signed data */
{
int32 i=sp->data_length/2;
int16 *tmp=(int16 *)sp->data;
while (i--)
*tmp++ ^= 0x8000;
}
/* Reverse reverse loops and pass them off as normal loops */
if (sp->modes & MODES_REVERSE)
{
int32 t;
/* The GUS apparently plays reverse loops by reversing the
whole sample. We do the same because the GUS does not SUCK. */
ctl->cmsg(CMSG_WARNING, VERB_NORMAL, "Reverse loop in %s", name);
reverse_data((int16 *)sp->data, 0, sp->data_length/2);
t=sp->loop_start;
sp->loop_start=sp->data_length - sp->loop_end;
sp->loop_end=sp->data_length - t;
sp->modes &= ~MODES_REVERSE;
sp->modes |= MODES_LOOPING; /* just in case */
}
/* If necessary do some anti-aliasing filtering */
if (antialiasing_allowed)
antialiasing(sp,play_mode->rate);
#ifdef ADJUST_SAMPLE_VOLUMES
if (amp!=-1)
sp->volume=(FLOAT_T)((amp) / 100.0);
else if (sf2flag)
sp->volume=(FLOAT_T)((sample_volume) / 255.0);
else
{
/* Try to determine a volume scaling factor for the sample.
This is a very crude adjustment, but things sound more
balanced with it. Still, this should be a runtime option. */
uint32 i, numsamps=sp->data_length/2;
uint32 higher=0, highcount=0;
int16 maxamp=0,a;
int16 *tmp=(int16 *)sp->data;
i = numsamps;
while (i--)
{
a=*tmp++;
if (a<0) a=-a;
if (a>maxamp)
maxamp=a;
}
tmp=(int16 *)sp->data;
i = numsamps;
while (i--)
{
a=*tmp++;
if (a<0) a=-a;
if (a > 3*maxamp/4)
{
higher += a;
highcount++;
}
}
if (highcount) higher /= highcount;
else higher = 10000;
sp->volume = (32768.0 * 0.875) / (double)higher ;
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * volume comp: %f", sp->volume);
}
#else
if (amp!=-1)
sp->volume=(double)(amp) / 100.0;
else
sp->volume=1.0;
#endif
sp->data_length /= 2; /* These are in bytes. Convert into samples. */
sp->loop_start /= 2;
sp->loop_end /= 2;
sp->data[sp->data_length] = sp->data[sp->data_length-1];
/* Then fractional samples */
sp->data_length <<= FRACTION_BITS;
sp->loop_start <<= FRACTION_BITS;
sp->loop_end <<= FRACTION_BITS;
/* trim off zero data at end */
{
int ls = sp->loop_start>>FRACTION_BITS;
int le = sp->loop_end>>FRACTION_BITS;
int se = sp->data_length>>FRACTION_BITS;
while (se > 1 && !sp->data[se-1]) se--;
if (le > se) le = se;
if (ls >= le) sp->modes &= ~MODES_LOOPING;
sp->loop_end = le<<FRACTION_BITS;
sp->data_length = se<<FRACTION_BITS;
}
/* Adjust for fractional loop points. This is a guess. Does anyone
know what "fractions" really stands for? */
sp->loop_start |=
(fractions & 0x0F) << (FRACTION_BITS-4);
sp->loop_end |=
((fractions>>4) & 0x0F) << (FRACTION_BITS-4);
/* If this instrument will always be played on the same note,
and it's not looped, we can resample it now. */
if (sp->note_to_use && !(sp->modes & MODES_LOOPING))
pre_resample(sp);
#ifdef LOOKUP_HACK
/* Squash the 16-bit data into 8 bits. */
{
uint8 *gulp,*ulp;
int16 *swp;
int l=sp->data_length >> FRACTION_BITS;
gulp=ulp=safe_malloc(l+1);
swp=(int16 *)sp->data;
while(l--)
*ulp++ = (*swp++ >> 8) & 0xFF;
free(sp->data);
sp->data=(sample_t *)gulp;
}
#endif
if (strip_tail==1)
{
/* Let's not really, just say we did. */
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Stripping tail");
sp->data_length = sp->loop_end;
}
} /* end of sample loop */
} /* end of stereo layer loop */
} /* end of vlayer loop */
close_file(fp);
return headlp;
}
int main(int argc, char **argv)
{
int32
note=DEFAULT_NOTE,
root=DEFAULT_ROOT,
verbosity=0,
samplerate=-1,
datasize=-1,
junklength, c;
int
infd=STDIN_FILENO;
unsigned char
thing[4096],
*point,
modes=0;
#ifdef __MACOS__
mac_main_wav2pat(&root, ¬e, &infd);
#else
while ((c=getopt(argc, argv, "r:n:vh"))>0)
switch(c)
{
case 'r': root=atol(optarg); break;
case 'n': root=freq_table[atoi(optarg) & 127]; break;
case 'u': note=atol(optarg); break;
case 'v': verbosity++; break;
case 'h': usage(); return 0;
default: return 1;
}
if (optind==argc-1)
{
if ((infd=open(argv[optind], O_RDONLY))<0)
{
perror("wav2pat: I can't open your WAVE");
return 3;
}
}
else if (optind!=argc)
{
usage();
return 1;
}
#endif
/* check out the putative RIFF WAVE on stdin */
if (read(infd, thing, 12)<=0)
{
perror("wav2pat: I can't read your WAVE");
return 3;
}
if (memcmp(thing, "RIFF", 4) || memcmp(thing+8, "WAVE", 4))
{
fprintf(stderr, "wav2pat: I want a RIFF WAVE on stdin!" NLS);
return 2;
}
while (datasize==-1)
{
if (read(infd, thing, 8)!=8)
{
perror("wav2pat: "
"Your WAVE ran out before I got to the interesting bits");
return 3;
}
junklength=LE_LONG(*((int32 *)(thing+4)));
/* This Microsoft file format is designed to be impossible to
parse correctly if one doesn't have the full specification.
If you have a wave with an INFO "chunk", you lose. Thank you
for playing. */
if (!memcmp(thing, "fmt ", 4))
{
if (junklength > 4096)
{
fprintf(stderr, "wav2pat: "
"WAVEs with %ld-byte format blocks make me throw up!"
NLS,
junklength);
return 2;
}
if (read(infd, thing, junklength) != junklength)
{
perror("wav2pat: Your WAVE is mangled");
return 3;
}
if (LE_SHORT(*((int16 *)(thing))) != 1)
{
fprintf(stderr, "wav2pat: I don't understand your WAVE. "
"It has a type %d format chunk!" NLS,
LE_SHORT(*((int16 *)(thing))));
return 2;
}
if (LE_SHORT(*((int16 *)(thing + 2))) != 1)
{
fprintf(stderr, "wav2pat: This WAVE has %d channels! "
"There can be only one!" NLS,
LE_SHORT(*((int16 *)(thing + 2))));
return 2;
}
samplerate=LE_LONG(*((int32 *)(thing + 4)));
switch(LE_SHORT(*((int16 *)(thing + 14))))
{
case 8: modes |= MODES_UNSIGNED; break;
case 16: modes |= MODES_16BIT; break;
default:
fprintf(stderr, "wav2pat: Ack! Ppthbth! %d-bit samples!",
LE_SHORT(*((int16 *)(thing + 14))));
return 2;
}
if (verbosity)
fprintf(stderr, "wav2pat: This is a %d-bit, %ld Hz WAVE" NLS,
(LE_SHORT(*((int16 *)(thing + 14)))),
samplerate);
}
else if (!memcmp(thing, "data", 4))
{
if (samplerate==-1)
{
fprintf(stderr, "wav2pat: Your WAVE has no format information before data!" NLS);
return 2;
}
if (verbosity)
fprintf(stderr, "wav2pat: It has %ld bytes of data" NLS, junklength);
datasize=junklength;
}
else
{
if (verbosity)
fprintf(stderr, "wav2pat: "
"Your WAVE has a %ld-byte chunk called `%4.4s'" NLS,
junklength, thing);
/* It's cool to pad chunks with NULs to align them on
half-word boundaries. */
if (junklength & 1)
junklength++;
while (junklength>0)
{
if ((c=read(infd, thing, (junklength>4096) ? 4096 : junklength))
<= 0)
{
perror("wav2pat: Now your WAVE has run out of data");
return 3;
}
junklength -= c;
}
}
}
/* hammer together something that looks like a GUS patch header */
#define pound(a) *point++=(a);
#define pounds(a) { int16 x = (int16)LE_SHORT(a); memcpy(point, &x, 2); point+=2; }
#define poundl(a) { int32 x = LE_LONG(a); memcpy(point, &x, 4); point+=4; }
#define bounce(a) point += a;
memset(thing, 0, 335);
point=thing;
/* header */
memcpy(point, "GF1PATCH110", 12);
point += 12;
/* Gravis ID */
memcpy(point, "ID#000002", 10);
point += 10;
/* description */
strcpy((char *)point, "Copyleft 1995 EWE&U Conductions and one Retreated Gravi\032");
point += 60;
pound(1); /* instruments */
pound(14); /* voices */
pound(0); /* channels */
pounds(1); /* waveforms */
pounds(127); /* master volume */
poundl(datasize); /* data size */
bounce(36); /* reserved */
pounds(1); /* instrument # */
strcpy((char *)point, "Bleahnoise"); /* instrument name */
point += 16;
poundl(datasize); /* instrument size */
pound(1); /* layers */
bounce(40); /* reserved */
pound(0); /* layer duplicate */
pound(0); /* layer */
poundl(datasize); /* layer size */
pound(1); /* samples */
bounce(40); /* reserved */
strcpy((char *)point, "bleah"); /* wave name */
point += 7;
pound(0); /* fractions */
poundl(datasize); /* wave size */
poundl(0); /* loop start */
poundl(datasize); /* loop end */
pounds(samplerate); /* sample rate */
poundl(8176); /* low freq */
poundl(12543854); /* high freq */
poundl(root); /* root freq */
pounds(512); /* tune */
pound(7); /* balance */
pound(63); /* envelope rates */
pound(63);
pound(63);
pound(63);
pound(63);
pound(63);
pound(240); /* envelope offsets */
pound(240);
pound(240);
pound(240);
pound(240);
pound(240);
pound(0); /* tremolo sweep */
pound(0); /* tremolo rate */
pound(0); /* tremolo depth */
pound(0); /* vibrato sweep */
pound(0); /* vibrato rate */
pound(0); /* vibrato depth */
pound(modes); /* modes */
pounds(note); /* scale freq */
pounds(1024); /* scale factor */
bounce(36); /* reserved */
write(STDOUT_FILENO, thing, 335);
/* wave data */
while (datasize>0)
{
if ((c=read(infd, thing, (datasize>4096) ? 4096 : datasize))
<= 0)
{
perror("wav2pat: I can't read data");
return 3;
}
write(STDOUT_FILENO, thing, c);
datasize -= c;
}
/* be courteous */
if (infd != STDIN_FILENO)
close(infd);
return 0;
}
/*
If panning or note_to_use != -1, it will be used for all samples,
instead of the sample-specific values in the instrument file.
For note_to_use, any value <0 or >127 will be forced to 0.
For other parameters, 1 means yes, 0 means no, other values are
undefined.
TODO: do reverse loops right */
static Instrument *load_instrument(char *name, int percussion,
int panning, int amp, int note_to_use,
int strip_loop, int strip_envelope,
int strip_tail)
{
ignore_unused_variable_warning(percussion);
Instrument *ip;
Sample *sp;
FILE *fp;
uint8 tmp[1024];
int i,j,noluck=0;
#ifdef PATCH_EXT_LIST
static const char *patch_ext[] = PATCH_EXT_LIST;
#endif
if (!name) return 0;
/* Open patch file */
if ((fp=open_file(name, 1, OF_NORMAL)) == NULL)
{
noluck=1;
#ifdef PATCH_EXT_LIST
/* Try with various extensions */
for (i=0; patch_ext[i]; i++)
{
if (strlen(name)+strlen(patch_ext[i])<1024)
{
char path[1024];
strcpy(path, name);
strcat(path, patch_ext[i]);
if ((fp=open_file(path, 1, OF_NORMAL)) != NULL)
{
noluck=0;
break;
}
}
}
#endif
}
if (noluck)
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Instrument `%s' can't be found.", name);
return 0;
}
ctl->cmsg(CMSG_INFO, VERB_NOISY, "Loading instrument %s", current_filename);
/* Read some headers and do cursory sanity checks. There are loads
of magic offsets. This could be rewritten... */
if ((239 != fread(tmp, 1, 239, fp)) ||
(memcmp(tmp, "GF1PATCH110\0ID#000002", 22) &&
memcmp(tmp, "GF1PATCH100\0ID#000002", 22))) /* don't know what the
differences are */
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "%s: not an instrument", name);
return 0;
}
if (tmp[82] != 1 && tmp[82] != 0) /* instruments. To some patch makers,
0 means 1 */
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Can't handle patches with %d instruments", tmp[82]);
return 0;
}
if (tmp[151] != 1 && tmp[151] != 0) /* layers. What's a layer? */
{
ctl->cmsg(CMSG_ERROR, VERB_NORMAL,
"Can't handle instruments with %d layers", tmp[151]);
return 0;
}
ip=safe_Malloc<Instrument>();
ip->samples = tmp[198];
ip->sample = safe_Malloc<Sample>(ip->samples);
for (i=0; i<ip->samples; i++)
{
uint8 fractions;
sint32 tmplong;
uint16 tmpshort;
uint8 tmpchar;
#define READ_CHAR(thing) \
if (1 != fread(&tmpchar, 1, 1, fp)) goto fail; \
thing = tmpchar;
#define READ_SHORT(thing) \
if (1 != fread(&tmpshort, 2, 1, fp)) goto fail; \
thing = LE_SHORT(tmpshort);
#define READ_LONG(thing) \
if (1 != fread(&tmplong, 4, 1, fp)) goto fail; \
thing = LE_LONG(tmplong);
skip(fp, 7); /* Skip the wave name */
if (1 != fread(&fractions, 1, 1, fp))
{
fail:
ctl->cmsg(CMSG_ERROR, VERB_NORMAL, "Error reading sample %d", i);
for (j=0; j<i; j++)
free(ip->sample[j].data);
free(ip->sample);
free(ip);
return 0;
}
sp=&(ip->sample[i]);
READ_LONG(sp->data_length);
READ_LONG(sp->loop_start);
READ_LONG(sp->loop_end);
READ_SHORT(sp->sample_rate);
READ_LONG(sp->low_freq);
READ_LONG(sp->high_freq);
READ_LONG(sp->root_freq);
skip(fp, 2); /* Why have a "root frequency" and then "tuning"?? */
READ_CHAR(tmp[0]);
if (panning==-1)
sp->panning = (tmp[0] * 8 + 4) & 0x7f;
else
sp->panning=static_cast<uint8>(panning & 0x7F);
/* envelope, tremolo, and vibrato */
if (18 != fread(tmp, 1, 18, fp)) goto fail;
if (!tmp[13] || !tmp[14])
{
sp->tremolo_sweep_increment=
sp->tremolo_phase_increment=sp->tremolo_depth=0;
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * no tremolo");
}
else
{
sp->tremolo_sweep_increment=convert_tremolo_sweep(tmp[12]);
sp->tremolo_phase_increment=convert_tremolo_rate(tmp[13]);
sp->tremolo_depth=tmp[14];
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" * tremolo: sweep %d, phase %d, depth %d",
sp->tremolo_sweep_increment, sp->tremolo_phase_increment,
sp->tremolo_depth);
}
if (!tmp[16] || !tmp[17])
{
sp->vibrato_sweep_increment=
sp->vibrato_control_ratio=sp->vibrato_depth=0;
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * no vibrato");
}
else
{
sp->vibrato_control_ratio=convert_vibrato_rate(tmp[16]);
sp->vibrato_sweep_increment=
convert_vibrato_sweep(tmp[15], sp->vibrato_control_ratio);
sp->vibrato_depth=tmp[17];
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" * vibrato: sweep %d, ctl %d, depth %d",
sp->vibrato_sweep_increment, sp->vibrato_control_ratio,
sp->vibrato_depth);
}
READ_CHAR(sp->modes);
skip(fp, 40); /* skip the useless scale frequency, scale factor
(what's it mean?), and reserved space */
/* Mark this as a fixed-pitch instrument if such a deed is desired. */
if (note_to_use!=-1)
sp->note_to_use=static_cast<uint8>(note_to_use);
else
sp->note_to_use=0;
/* seashore.pat in the Midia patch set has no Sustain. I don't
understand why, and fixing it by adding the Sustain flag to
all looped patches probably breaks something else. We do it
anyway. */
if (sp->modes & MODES_LOOPING)
sp->modes |= MODES_SUSTAIN;
/* Strip any loops and envelopes we're permitted to */
if ((strip_loop==1) &&
(sp->modes & (MODES_SUSTAIN | MODES_LOOPING |
MODES_PINGPONG | MODES_REVERSE)))
{
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Removing loop and/or sustain");
sp->modes &=~(MODES_SUSTAIN | MODES_LOOPING |
MODES_PINGPONG | MODES_REVERSE);
}
if (strip_envelope==1)
{
if (sp->modes & MODES_ENVELOPE)
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Removing envelope");
sp->modes &= ~MODES_ENVELOPE;
}
else if (strip_envelope != 0)
{
/* Have to make a guess. */
if (!(sp->modes & (MODES_LOOPING | MODES_PINGPONG | MODES_REVERSE)))
{
/* No loop? Then what's there to sustain? No envelope needed
either... */
sp->modes &= ~(MODES_SUSTAIN|MODES_ENVELOPE);
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - No loop, removing sustain and envelope");
}
else if (!memcmp(tmp, "??????", 6) || tmp[11] >= 100)
{
/* Envelope rates all maxed out? Envelope end at a high "offset"?
That's a weird envelope. Take it out. */
sp->modes &= ~MODES_ENVELOPE;
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - Weirdness, removing envelope");
}
else if (!(sp->modes & MODES_SUSTAIN))
{
/* No sustain? Then no envelope. I don't know if this is
justified, but patches without sustain usually don't need the
envelope either... at least the Gravis ones. They're mostly
drums. I think. */
sp->modes &= ~MODES_ENVELOPE;
ctl->cmsg(CMSG_INFO, VERB_DEBUG,
" - No sustain, removing envelope");
}
}
for (j=0; j<6; j++)
{
sp->envelope_rate[j]=
convert_envelope_rate(tmp[j]);
sp->envelope_offset[j]=
convert_envelope_offset(tmp[6+j]);
}
/* Then read the sample data */
sp->data = safe_Malloc<sample_t>(sp->data_length);
if (1 != fread(sp->data, sp->data_length, 1, fp))
goto fail;
if (!(sp->modes & MODES_16BIT)) /* convert to 16-bit data */
{
sint32 i=sp->data_length;
uint8 *cp=reinterpret_cast<uint8 *>(sp->data);
uint16 *tmp,*new_dat;
tmp=new_dat=safe_Malloc<uint16>(sp->data_length);
while (i--)
*tmp++ = static_cast<uint16>(*cp++) << 8;
cp=reinterpret_cast<uint8 *>(sp->data);
sp->data = reinterpret_cast<sample_t *>(new_dat);
free(cp);
sp->data_length *= 2;
sp->loop_start *= 2;
sp->loop_end *= 2;
}
#ifndef TIMIDITY_LITTLE_ENDIAN
else
/* convert to machine byte order */
{
sint32 i=sp->data_length/2;
sint16 *tmp=reinterpret_cast<sint16 *>(sp->data),s;
while (i--)
{
s=LE_SHORT(*tmp);
*tmp++=s;
}
}
#endif
if (sp->modes & MODES_UNSIGNED) /* convert to signed data */
{
sint32 i=sp->data_length/2;
sint16 *tmp = sp->data;
while (i--)
*tmp++ ^= 0x8000;
}
/* Reverse reverse loops and pass them off as normal loops */
if (sp->modes & MODES_REVERSE)
{
sint32 t;
/* The GUS apparently plays reverse loops by reversing the
whole sample. We do the same because the GUS does not SUCK. */
ctl->cmsg(CMSG_WARNING, VERB_NORMAL, "Reverse loop in %s", name);
reverse_data(sp->data, 0, sp->data_length/2);
t=sp->loop_start;
sp->loop_start=sp->data_length - sp->loop_end;
sp->loop_end=sp->data_length - t;
sp->modes &= ~MODES_REVERSE;
sp->modes |= MODES_LOOPING; /* just in case */
}
/* If necessary do some anti-aliasing filtering */
if (antialiasing_allowed)
antialiasing(sp,play_mode->rate);
#ifdef ADJUST_SAMPLE_VOLUMES
if (amp!=-1)
sp->volume=static_cast<float>((amp) / 100.0);
else
{
/* Try to determine a volume scaling factor for the sample.
This is a very crude adjustment, but things sound more
balanced with it. Still, this should be a runtime option. */
sint32 i=sp->data_length/2;
sint16 maxamp=0,a;
sint16 *tmp = sp->data;
while (i--)
{
a=*tmp++;
if (a<0) a=-a;
if (a>maxamp)
maxamp=a;
}
sp->volume=static_cast<float>(32768.0 / maxamp);
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " * volume comp: %f", sp->volume);
}
#else
if (amp!=-1)
sp->volume=static_cast<double>(amp) / 100.0;
else
sp->volume=1.0;
#endif
sp->data_length /= 2; /* These are in bytes. Convert into samples. */
sp->loop_start /= 2;
sp->loop_end /= 2;
/* Then fractional samples */
sp->data_length <<= FRACTION_BITS;
sp->loop_start <<= FRACTION_BITS;
sp->loop_end <<= FRACTION_BITS;
/* Adjust for fractional loop points. This is a guess. Does anyone
know what "fractions" really stands for? */
sp->loop_start |=
(fractions & 0x0F) << (FRACTION_BITS-4);
sp->loop_end |=
((fractions>>4) & 0x0F) << (FRACTION_BITS-4);
/* If this instrument will always be played on the same note,
and it's not looped, we can resample it now. */
if (sp->note_to_use && !(sp->modes & MODES_LOOPING))
pre_resample(sp);
#ifdef LOOKUP_HACK
/* Squash the 16-bit data into 8 bits. */
{
uint8 *gulp,*ulp;
sint16 *swp;
int l=sp->data_length >> FRACTION_BITS;
gulp=ulp=safe_Malloc<uint8>(l+1);
swp=(sint16 *)sp->data;
while(l--)
*ulp++ = (*swp++ >> 8) & 0xFF;
free(sp->data);
sp->data=(sample_t *)gulp;
}
#endif
if (strip_tail==1)
{
/* Let's not really, just say we did. */
ctl->cmsg(CMSG_INFO, VERB_DEBUG, " - Stripping tail");
sp->data_length = sp->loop_end;
}
}
close_file(fp);
return ip;
}
int SNDWAV_SetParams(SNDPCMContainer_t *sndpcm, WAVContainer_t *wav)
{
snd_pcm_hw_params_t *hwparams;
snd_pcm_format_t format;
uint32_t exact_rate;
uint32_t buffer_time, period_time;
/* Allocate the snd_pcm_hw_params_t structure on the stack. */
snd_pcm_hw_params_alloca(&hwparams);
/* Init hwparams with full configuration space */
if (snd_pcm_hw_params_any(sndpcm->handle, hwparams) < 0) {
fprintf(stderr, "Error snd_pcm_hw_params_any/n");
goto ERR_SET_PARAMS;
}
if (snd_pcm_hw_params_set_access(sndpcm->handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
fprintf(stderr, "Error snd_pcm_hw_params_set_access\n");
goto ERR_SET_PARAMS;
}
/* Set sample format */
if (SNDWAV_P_GetFormat(wav, &format) < 0) {
fprintf(stderr, "Error get_snd_pcm_format\n");
goto ERR_SET_PARAMS;
}
if (snd_pcm_hw_params_set_format(sndpcm->handle, hwparams, format) < 0) {
fprintf(stderr, "Error snd_pcm_hw_params_set_format\n");
goto ERR_SET_PARAMS;
}
sndpcm->format = format;
/* Set number of channels */
if (snd_pcm_hw_params_set_channels(sndpcm->handle, hwparams, LE_SHORT(wav->format.channels)) < 0) {
fprintf(stderr, "Error snd_pcm_hw_params_set_channels\n");
goto ERR_SET_PARAMS;
}
sndpcm->channels = LE_SHORT(wav->format.channels);
/* Set sample rate. If the exact rate is not supported */
/* by the hardware, use nearest possible rate. */
exact_rate = LE_INT(wav->format.sample_rate);
if (snd_pcm_hw_params_set_rate_near(sndpcm->handle, hwparams, &exact_rate, 0) < 0) {
fprintf(stderr, "Error snd_pcm_hw_params_set_rate_near\n");
goto ERR_SET_PARAMS;
}
if (LE_INT(wav->format.sample_rate) != exact_rate) {
fprintf(stderr, "The rate %d Hz is not supported by your hardware.\n ==> Using %d Hz instead.\n",
LE_INT(wav->format.sample_rate), exact_rate);
}
if (snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time, 0) < 0) {
fprintf(stderr, "Error snd_pcm_hw_params_get_buffer_time_max\n");
goto ERR_SET_PARAMS;
}
if (buffer_time > 500000) buffer_time = 500000;
period_time = buffer_time / 4;
if (snd_pcm_hw_params_set_buffer_time_near(sndpcm->handle, hwparams, &buffer_time, 0) < 0) {
fprintf(stderr, "Error snd_pcm_hw_params_set_buffer_time_near\n");
goto ERR_SET_PARAMS;
}
if (snd_pcm_hw_params_set_period_time_near(sndpcm->handle, hwparams, &period_time, 0) < 0) {
fprintf(stderr, "Error snd_pcm_hw_params_set_period_time_near\n");
goto ERR_SET_PARAMS;
}
/* Set hw params */
if (snd_pcm_hw_params(sndpcm->handle, hwparams) < 0) {
fprintf(stderr, "Error snd_pcm_hw_params(handle, params)\n");
goto ERR_SET_PARAMS;
}
snd_pcm_hw_params_get_period_size(hwparams, &sndpcm->chunk_size, 0);
snd_pcm_hw_params_get_buffer_size(hwparams, &sndpcm->buffer_size);
if (sndpcm->chunk_size == sndpcm->buffer_size) {
fprintf(stderr, ("Can't use period equal to buffer size (%lu == %lu)\n"), sndpcm->chunk_size, sndpcm->buffer_size);
goto ERR_SET_PARAMS;
}
sndpcm->bits_per_sample = snd_pcm_format_physical_width(format);
sndpcm->bits_per_frame = sndpcm->bits_per_sample * LE_SHORT(wav->format.channels);
sndpcm->chunk_bytes = sndpcm->chunk_size * sndpcm->bits_per_frame / 8;
/* Allocate audio data buffer */
sndpcm->data_buf = (uint8_t *)malloc(sndpcm->chunk_bytes);
if (!sndpcm->data_buf) {
fprintf(stderr, "Error malloc: [data_buf]\n");
goto ERR_SET_PARAMS;
}
return 0;
ERR_SET_PARAMS:
return -1;
}
