/**
* <JA>
* 音声入力デバイスを初期化し，音入力切出用パラメータをセットアップする.
*
* @param adin [in] AD-in ワークエリア
* @param jconf [in] 全体設定パラメータ
* @param arg [in] デバイス依存引数
* </JA>
* <EN>
* Initialize audio device and set up parameters for sound detection.
*
* @param adin [in] AD-in work area
* @param jconf [in] global configuration parameters
* @param arg [in] device-specific argument
* </EN>
*/
static boolean
adin_setup_all(ADIn *adin, Jconf *jconf, void *arg)
{
if (jconf->input.use_ds48to16) {
if (jconf->input.use_ds48to16 && jconf->input.sfreq != 16000) {
jlog("ERROR: m_adin: in 48kHz input mode, target sampling rate should be 16k!\n");
return FALSE;
}
/* setup for 1/3 down sampling */
adin->ds = ds48to16_new();
adin->down_sample = TRUE;
/* set device sampling rate to 48kHz */
if (adin_standby(adin, 48000, arg) == FALSE) { /* fail */
jlog("ERROR: m_adin: failed to ready input device\n");
return FALSE;
}
} else {
adin->ds = NULL;
adin->down_sample = FALSE;
if (adin_standby(adin, jconf->input.sfreq, arg) == FALSE) { /* fail */
jlog("ERROR: m_adin: failed to ready input device\n");
return FALSE;
}
}
/* set parameter for recording/silence detection */
if (adin_setup_param(adin, jconf) == FALSE) {
jlog("ERROR: m_adin: failed to set parameter for input device\n");
return FALSE;
}
adin->input_side_segment = FALSE;
return TRUE;
}
/**
* <JA>
* メイン関数
*
* @param argc [in] 引数列の長さ
* @param argv [in] 引数列
*
* @return
* </JA>エラー時 1，通常終了時 0 を返す．
* <EN>
* Main function.
*
* @param argc [in] number of argument.
* @param argv [in] array of arguments.
*
* @return 1 on error, 0 on success.
* </EN>
*/
int
main(int argc, char *argv[])
{
int silence_cut = 1; /* 0=disabled, 1=enabled, 2=undef */
/* parse option */
opt_parse(argc, argv);
/* select microphone input */
if (adin_select(SP_MIC) == FALSE) {
j_printerr("Error: microphone input is not supported\n");
return(1);
}
/* ready the microphone (arg is dummy) */
if (use_48to16) {
if (sfreq != 16000) {
j_printerr("Error: in 48kHz mode, required rate should be 16kHz!\n");
return(1);
}
/* setup for down sampling */
adin_setup_48to16();
/* set device sampling rate to 48kHz */
if (adin_standby(48000, NULL) == FALSE) { /* fail */
j_printerr("Error: failed to ready input device to 48kHz\n");
return(1);
}
} else {
if (adin_standby(sfreq, NULL) == FALSE) {
j_printerr("Error: failed to standby input\n");
return(1);
}
}
/* set device-independent param */
adin_setup_param(silence_cut, strip_zero_sample, level_thres, zero_cross_num, margin, margin, sfreq, TRUE, use_zmean);/* use same margin for head and tail */
/* file check */
if (!stout) {
if (access(filename, F_OK) == 0) {
if (access(filename, W_OK) == 0) {
fprintf(stderr, "Warning: overwriting file \"%s\"\n", filename);
} else {
perror("adinrec");
return(1);
}
}
} /* output file will be opened when triggered (not here) */
/* set interrupt signal handler to properly close output file */
if (signal(SIGINT, interrupt_record) == SIG_ERR) {
fprintf(stderr, "Warning: signal intterupt may collapse output\n");
}
/* do recoding */
speechlen = 0;
adin_begin();
j_printerr("<<< please speak >>>"); /* moved from adin-cut.c */
adin_go(adin_callback_file, NULL);
adin_end();
/* close file and output status */
close_file();
return 1;
}
//========================
// Create engine instance
//========================
bool cJulius::createEngine( void )
{
#ifdef APP_ADIN
ADIn *a;
#endif
if (!m_jconf) return false;
if (m_recog) return false;
#ifdef APP_ADIN
if (m_appsource != 0) {
switch(m_appsource) {
case 1: // buffer input, batch
m_recog->jconf->input.type = INPUT_WAVEFORM;
m_recog->jconf->input.speech_input = SP_RAWFILE;
m_recog->jconf->decodeopt.realtime_flag = FALSE;
break;
case 2: // buffer input, incremental
m_recog->jconf->input.type = INPUT_WAVEFORM;
m_recog->jconf->input.speech_input = SP_RAWFILE;
m_recog->jconf->decodeopt.realtime_flag = TRUE;
break;
}
}
#endif
// Create engine instance
m_recog = j_create_instance_from_jconf(m_jconf);
if (m_recog == NULL) {
return false;
}
// Register callbacks
callback_add(m_recog, CALLBACK_EVENT_PROCESS_ONLINE, ::callback_engine_active, this);
callback_add(m_recog, CALLBACK_EVENT_PROCESS_OFFLINE, ::callback_engine_inactive, this);
callback_add(m_recog, CALLBACK_EVENT_SPEECH_READY, ::callback_audio_ready, this);
callback_add(m_recog, CALLBACK_EVENT_SPEECH_START, ::callback_audio_begin, this);
callback_add(m_recog, CALLBACK_EVENT_SPEECH_STOP, ::callback_audio_end, this);
callback_add(m_recog, CALLBACK_EVENT_RECOGNITION_BEGIN, ::callback_recog_begin, this);
callback_add(m_recog, CALLBACK_EVENT_RECOGNITION_END, ::callback_recog_end, this);
callback_add(m_recog, CALLBACK_EVENT_PASS1_FRAME, ::callback_recog_frame, this);
callback_add(m_recog, CALLBACK_EVENT_PAUSE, ::callback_engine_pause, this);
callback_add(m_recog, CALLBACK_EVENT_RESUME, ::callback_engine_resume, this);
callback_add(m_recog, CALLBACK_RESULT, ::callback_result_final, this);
callback_add(m_recog, CALLBACK_PAUSE_FUNCTION, ::callback_wait_for_resume, this);
#ifdef APP_ADIN
// Initialize application side audio input
if (m_appsource != 0) {
a = m_recog->adin;
switch(m_appsource) {
case 1: // buffer input, batch
a->ad_standby = NULL;
a->ad_begin = NULL;
a->ad_end = NULL;
a->ad_resume = NULL;
a->ad_pause = NULL;
a->ad_terminate = NULL;
a->ad_read = callback_adin_fetch_input;
a->ad_input_name = NULL;
a->silence_cut_default = FALSE;
a->enable_thread = FALSE;
break;
case 2: // buffer input, incremental
a->ad_standby = NULL;
a->ad_begin = NULL;
a->ad_end = NULL;
a->ad_resume = NULL;
a->ad_pause = NULL;
a->ad_terminate = NULL;
a->ad_read = callback_adin_fetch_input;
a->ad_input_name = NULL;
a->silence_cut_default = FALSE;
a->enable_thread = FALSE;
break;
}
a->ds = NULL;
a->down_sample = FALSE;
if (adin_standby(a, m_recog->jconf->input.sfreq, NULL) == FALSE) return false;
if (adin_setup_param(a, m_recog->jconf) == FALSE) return false;
a->input_side_segment = FALSE;
} else {
// Let JuliusLib get audio input
if (! j_adin_init( m_recog ) ) return false;
}
#else
if (! j_adin_init( m_recog ) ) return false;
#endif
return true;
}
