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/* ----------------------------------------------------------------- */
/* The HMM-Based Speech Synthesis Engine "hts_engine API" */
/* developed by HTS Working Group */
/* http://hts-engine.sourceforge.net/ */
/* ----------------------------------------------------------------- */
/* */
/* Copyright (c) 2001-2015 Nagoya Institute of Technology */
/* Department of Computer Science */
/* */
/* 2001-2008 Tokyo Institute of Technology */
/* Interdisciplinary Graduate School of */
/* Science and Engineering */
/* */
/* All rights reserved. */
/* */
/* Redistribution and use in source and binary forms, with or */
/* without modification, are permitted provided that the following */
/* conditions are met: */
/* */
/* - Redistributions of source code must retain the above copyright */
/* notice, this list of conditions and the following disclaimer. */
/* - Redistributions in binary form must reproduce the above */
/* copyright notice, this list of conditions and the following */
/* disclaimer in the documentation and/or other materials provided */
/* with the distribution. */
/* - Neither the name of the HTS working group nor the names of its */
/* contributors may be used to endorse or promote products derived */
/* from this software without specific prior written permission. */
/* */
/* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND */
/* CONTRIBUTORS "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, */
/* INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF */
/* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE */
/* DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS */
/* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, */
/* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED */
/* TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, */
/* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON */
/* ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, */
/* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY */
/* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE */
/* POSSIBILITY OF SUCH DAMAGE. */
/* ----------------------------------------------------------------- */
#ifndef HTS_GSTREAM_C
#define HTS_GSTREAM_C
#ifdef __cplusplus
#define HTS_GSTREAM_C_START extern "C" {
#define HTS_GSTREAM_C_END }
#else
#define HTS_GSTREAM_C_START
#define HTS_GSTREAM_C_END
#endif /* __CPLUSPLUS */
HTS_GSTREAM_C_START;
/* hts_engine libraries */
#include "HTS_hidden.h"
/* HTS_GStreamSet_initialize: initialize generated parameter stream set */
void HTS_GStreamSet_initialize(HTS_GStreamSet * gss)
{
gss->nstream = 0;
gss->total_frame = 0;
gss->total_nsample = 0;
gss->gstream = NULL;
gss->gspeech = NULL;
}
/* 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) == TRUE) {
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;
}
/* HTS_GStreamSet_get_total_nsamples: get total number of sample */
size_t HTS_GStreamSet_get_total_nsamples(HTS_GStreamSet * gss)
{
return gss->total_nsample;
}
/* HTS_GStreamSet_get_total_frame: get total number of frame */
size_t HTS_GStreamSet_get_total_frame(HTS_GStreamSet * gss)
{
return gss->total_frame;
}
/* HTS_GStreamSet_get_vector_length: get features length */
size_t HTS_GStreamSet_get_vector_length(HTS_GStreamSet * gss, size_t stream_index)
{
return gss->gstream[stream_index].vector_length;
}
/* HTS_GStreamSet_get_speech: get synthesized speech parameter */
double HTS_GStreamSet_get_speech(HTS_GStreamSet * gss, size_t sample_index)
{
return gss->gspeech[sample_index];
}
/* HTS_GStreamSet_get_parameter: get generated parameter */
double HTS_GStreamSet_get_parameter(HTS_GStreamSet * gss, size_t stream_index, size_t frame_index, size_t vector_index)
{
return gss->gstream[stream_index].par[frame_index][vector_index];
}
/* HTS_GStreamSet_clear: free generated parameter stream set */
void HTS_GStreamSet_clear(HTS_GStreamSet * gss)
{
size_t i, j;
if (gss->gstream) {
for (i = 0; i < gss->nstream; i++) {
if (gss->gstream[i].par != NULL) {
for (j = 0; j < gss->total_frame; j++)
HTS_free(gss->gstream[i].par[j]);
HTS_free(gss->gstream[i].par);
}
}
HTS_free(gss->gstream);
}
if (gss->gspeech)
HTS_free(gss->gspeech);
HTS_GStreamSet_initialize(gss);
}
HTS_GSTREAM_C_END;
#endif /* !HTS_GSTREAM_C */
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