/* HTS_Audio_clear: free audio */
void HTS_Audio_clear(HTS_Audio * audio)
{
HTS_Audio_flush(audio);
HTS_Audio_close(audio);
}
/* HTS_GStreamSet_create: generate speech */
HTS_Boolean HTS_GStreamSet_create(HTS_GStreamSet * gss, HTS_PStreamSet * pss, size_t stage, HTS_Boolean use_log_gain, size_t sampling_rate, size_t fperiod, double alpha, double beta, HTS_Boolean * stop, double volume, HTS_Audio * audio)
{
size_t i, j, k;
size_t msd_frame;
HTS_Vocoder v;
size_t nlpf = 0;
double *lpf = NULL;
/* check */
if (gss->gstream || gss->gspeech) {
HTS_error(1, "HTS_GStreamSet_create: HTS_GStreamSet is not initialized.\n");
return FALSE;
}
/* initialize */
gss->nstream = HTS_PStreamSet_get_nstream(pss);
gss->total_frame = HTS_PStreamSet_get_total_frame(pss);
gss->total_nsample = fperiod * gss->total_frame;
gss->gstream = (HTS_GStream *) HTS_calloc(gss->nstream, sizeof(HTS_GStream));
for (i = 0; i < gss->nstream; i++) {
gss->gstream[i].vector_length = HTS_PStreamSet_get_vector_length(pss, i);
gss->gstream[i].par = (double **) HTS_calloc(gss->total_frame, sizeof(double *));
for (j = 0; j < gss->total_frame; j++)
gss->gstream[i].par[j] = (double *) HTS_calloc(gss->gstream[i].vector_length, sizeof(double));
}
gss->gspeech = (double *) HTS_calloc(gss->total_nsample, sizeof(double));
/* copy generated parameter */
for (i = 0; i < gss->nstream; i++) {
if (HTS_PStreamSet_is_msd(pss, i)) { /* for MSD */
for (j = 0, msd_frame = 0; j < gss->total_frame; j++)
if (HTS_PStreamSet_get_msd_flag(pss, i, j)) {
for (k = 0; k < gss->gstream[i].vector_length; k++)
gss->gstream[i].par[j][k] = HTS_PStreamSet_get_parameter(pss, i, msd_frame, k);
msd_frame++;
} else
for (k = 0; k < gss->gstream[i].vector_length; k++)
gss->gstream[i].par[j][k] = HTS_NODATA;
} else { /* for non MSD */
for (j = 0; j < gss->total_frame; j++)
for (k = 0; k < gss->gstream[i].vector_length; k++)
gss->gstream[i].par[j][k] = HTS_PStreamSet_get_parameter(pss, i, j, k);
}
}
/* check */
if (gss->nstream != 2 && gss->nstream != 3) {
HTS_error(1, "HTS_GStreamSet_create: The number of streams should be 2 or 3.\n");
HTS_GStreamSet_clear(gss);
return FALSE;
}
if (HTS_PStreamSet_get_vector_length(pss, 1) != 1) {
HTS_error(1, "HTS_GStreamSet_create: The size of lf0 static vector should be 1.\n");
HTS_GStreamSet_clear(gss);
return FALSE;
}
if (gss->nstream >= 3 && gss->gstream[2].vector_length % 2 == 0) {
HTS_error(1, "HTS_GStreamSet_create: The number of low-pass filter coefficient should be odd numbers.");
HTS_GStreamSet_clear(gss);
return FALSE;
}
/* synthesize speech waveform */
HTS_Vocoder_initialize(&v, gss->gstream[0].vector_length - 1, stage, use_log_gain, sampling_rate, fperiod);
if (gss->nstream >= 3)
nlpf = gss->gstream[2].vector_length;
for (i = 0; i < gss->total_frame && (*stop) == FALSE; i++) {
j = i * fperiod;
if (gss->nstream >= 3)
lpf = &gss->gstream[2].par[i][0];
HTS_Vocoder_synthesize(&v, gss->gstream[0].vector_length - 1, gss->gstream[1].par[i][0], &gss->gstream[0].par[i][0], nlpf, lpf, alpha, beta, volume, &gss->gspeech[j], audio);
}
HTS_Vocoder_clear(&v);
if (audio)
HTS_Audio_flush(audio);
return TRUE;
}