/* ----------------------------------------------------------------- */ /* 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 */