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speech-tools/sigpr/pda/pda.cc

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[DEV] 2.4 tools
2015-09-19 10:52:26 +02:00
/*************************************************************************/
/* */
/* Centre for Speech Technology Research */
/* University of Edinburgh, UK */
/* Copyright (c) 1995,1996 */
/* All Rights Reserved. */
/* */
/* Permission is hereby granted, free of charge, to use and distribute */
/* this software and its documentation without restriction, including */
/* without limitation the rights to use, copy, modify, merge, publish, */
/* distribute, sublicense, and/or sell copies of this work, and to */
/* permit persons to whom this work is furnished to do so, subject to */
/* the following conditions: */
/* 1. The code must retain the above copyright notice, this list of */
/* conditions and the following disclaimer. */
/* 2. Any modifications must be clearly marked as such. */
/* 3. Original authors' names are not deleted. */
/* 4. The authors' names are not used to endorse or promote products */
/* derived from this software without specific prior written */
/* permission. */
/* */
/* THE UNIVERSITY OF EDINBURGH AND THE CONTRIBUTORS TO THIS WORK */
/* DISCLAIM ALL WARRANTIES WITH REGARD TO THIS SOFTWARE, INCLUDING */
/* ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS, IN NO EVENT */
/* SHALL THE UNIVERSITY OF EDINBURGH NOR THE CONTRIBUTORS BE LIABLE */
/* FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES */
/* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN */
/* AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, */
/* ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF */
/* THIS SOFTWARE. */
/* */
/*************************************************************************/
/* Author : Paul Taylor */
/* Date : April 1994 */
/*************************************************************************/
#include "EST_speech_class.h"
#include "sigpr/EST_sigpr_utt.h"
#include "sigpr/EST_filter.h"
#include "srpd.h"
#include "EST_error.h"
#include "EST_string_aux.h"
int read_next_wave_segment (EST_Wave &sig, struct Srpd_Op *paras,
SEGMENT_ *p_seg);
static void srpd(EST_Wave &sig, EST_Track &fz, Srpd_Op &srpd_op, int resize);
static struct Srpd_Op *default_srpd_op(struct Srpd_Op *srpd);
static void parse_srpd_list(EST_Features &a_list, struct Srpd_Op *srpd);
void pda(EST_Wave &sig, EST_Track &fz, EST_Features &op, EST_String method)
{
if (method == "")
{
if (op.present("pda_method"))
method = op.S("pda_method");
}
if (method == "")
srpd(sig, fz, op);
else if (method == "srpd")
srpd(sig, fz, op);
else
EST_error("Unknown pda %s\n", (const char *)method);
}
void icda(EST_Wave &sig, EST_Track &fz, EST_Track &speech, EST_Features &op,
EST_String method)
{ // intonation contour detection algorithm
EST_Track raw_fz;
if (method == "")
{
if (op.present("pda_method"))
method = op.S("pda_method");
}
if (method == "")
srpd(sig, raw_fz, op);
else if (method == "srpd")
srpd(sig, raw_fz, op);
else
EST_error("Unknown pda %s\n", (const char *)method);
smooth_phrase(raw_fz, speech, op, fz);
}
void srpd(EST_Wave &sig, EST_Track &fz, EST_Features &op)
{
Srpd_Op srpd_op;
default_srpd_op(&srpd_op); // default values
parse_srpd_list(op, &srpd_op); // override with options
if (op.I("do_low_pass",0))
FIRlowpass_filter(sig, op.I("lpf_cutoff"),op.I("lpf_order"));
srpd(sig, fz, srpd_op, op.I("srpd_resize", 0));
}
/*void do_srpd_fz(EST_Wave &sig, EST_Track &fz)
{
Srpd_Op srpd_op;
default_srpd_op(&srpd_op);
srpd(sig, fz, srpd_op, 1);
}
*/
void srpd(EST_Wave &sig, EST_Track &fz, Srpd_Op &srpd_op, int resize)
{
int i, rns, tracklen, j = 0;
SEGMENT_ segment;
CROSS_CORR_ cc;
STATUS_ pda_status, held_status;
srpd_op.sample_freq = sig.sample_rate();
float min, max;
min = srpd_op.min_pitch; // must store as set up routines corrupt
max = srpd_op.max_pitch;
initialise_structures (&srpd_op, &segment, &cc);
initialise_status (&srpd_op, &pda_status);
initialise_status (&srpd_op, &held_status);
tracklen = (sig.num_samples() - segment.length) / segment.shift + 1;
if (resize)
{
fz.set_equal_space(true);
fz.resize(tracklen, 1);
fz.set_channel_name("F0", 0);
fz.fill_time(srpd_op.shift/1000);
}
if (!fz.equal_space())
EST_error("Pitch tracking algorithm must have equal spaced track\n");
while ((rns = read_next_wave_segment (sig, &srpd_op, &segment)) != 0)
{
if (rns == 2)
{
for (i = 0; i < cc.size; cc.coeff[i++] = 0.0);
initialise_status (&srpd_op, &pda_status);
}
else
super_resolution_pda (&srpd_op, segment, &cc, &pda_status);
if (pda_status.s_h == HOLD)
{
held_status.pitch_freq = pda_status.pitch_freq;
held_status.v_uv = VOICED;
held_status.s_h = HELD;
held_status.cc_max = pda_status.cc_max;
held_status.threshold = pda_status.threshold;
continue;
}
if (held_status.s_h == HELD)
{
if (pda_status.pitch_freq == BREAK_NUMBER)
{
held_status.pitch_freq = BREAK_NUMBER;
held_status.v_uv = UNVOICED;
}
held_status.s_h = SENT;
if (held_status.v_uv != VOICED)
fz.set_break(j);
fz.a(j++) = held_status.pitch_freq;
// printf( "track set: %d (of %d) to %f\n", j-1, fz.length(), held_status.pitch_freq );
}
if (pda_status.v_uv != VOICED)
fz.set_break(j);
fz.a(j++) = pda_status.pitch_freq;
//printf( "track set: %d (of %d) to %f\n", j-1, fz.length(), pda_status.pitch_freq );
}
if (held_status.s_h == HELD)
{
held_status.pitch_freq = BREAK_NUMBER;
held_status.v_uv = UNVOICED;
fz.set_break(j);
fz.a(j++) = held_status.pitch_freq;
}
end_structure_use (&segment, &cc);
}
static struct Srpd_Op *default_srpd_op(struct Srpd_Op *srpd)
{
srpd->L = DEFAULT_DECIMATION;
srpd->min_pitch = DEFAULT_MIN_PITCH;
srpd->max_pitch = DEFAULT_MAX_PITCH;
srpd->shift = DEFAULT_SHIFT;
srpd->length = DEFAULT_LENGTH;
srpd->Tsilent = DEFAULT_TSILENT;
srpd->Tmin = DEFAULT_TMIN;
srpd->Tmax_ratio = DEFAULT_TMAX_RATIO;
srpd->Thigh = DEFAULT_THIGH;
srpd->Tdh = DEFAULT_TDH;
srpd->make_ascii = 0;
srpd->peak_tracking = 0;
srpd->sample_freq = DEFAULT_SF;
/* p_par->Nmax and p_par->Nmin cannot be initialised */
return(srpd);
}
static void parse_srpd_list(EST_Features &al, struct Srpd_Op *srpd)
{
if (al.present("decimation"))
srpd->L = al.I("decimation");
if (al.present("min_pitch"))
srpd->min_pitch = al.F("min_pitch");
if (al.present("max_pitch"))
srpd->max_pitch = al.F("max_pitch");
if (al.present("pda_frame_shift"))
srpd->shift = al.F("pda_frame_shift") * 1000.0;
if (al.present("pda_frame_length"))
srpd->length = al.F("pda_frame_length") * 1000.0;
if (al.present("noise_floor"))
srpd->Tsilent = al.I("noise_floor");
if (al.present("v2uv_coeff_thresh"))
srpd->Thigh = al.F("v2uv_coef_thresh");
if (al.present("min_v2uv_coef_thresh"))
srpd->Tmin = al.F("min_v2uv_coef_thresh");
if (al.present("v2uv_coef_thresh_ratio"))
srpd->Tmax_ratio = al.F("v2uv_coef_thresh_ratio");
if (al.present("anti_doubling_thresh"))
srpd->Tdh = al.F("anti_doubling_thresh");
if (al.present("peak_tracking"))
srpd->peak_tracking = al.I("peak_tracking");
if (al.present("sample_frequency"))
srpd->sample_freq = al.I("sample_frequency");
}
void default_pda_options(EST_Features &al)
{
al.set("min_pitch", "40.0");
al.set("max_pitch", "400.0");
al.set("pda_frame_shift", "0.005");
al.set("pda_frame_length", DEFAULT_LENGTH / 1000.0);
al.set("lpf_cutoff", "600");
al.set("lpf_order", "49");
al.set("f0_file_type", "esps");
al.set("decimation", DEFAULT_DECIMATION);
al.set("noise_floor", DEFAULT_TSILENT);
al.set("min_v2uv_coef_thresh", DEFAULT_TMIN);
al.set("v2uv_coef_thresh_ratio", DEFAULT_TMAX_RATIO);
al.set("v2uv_coef_thresh", DEFAULT_THIGH);
al.set("anti_doubling_thresh", DEFAULT_TDH);
al.set("peak_tracking", 0);
}
EST_String options_pda_general(void)
{
// The standard waveform input options
return
EST_String("")+
"-L Perform low pass filtering on input. This option should always \n"
" be used in normal processing as it usually increases \n"
" performance considerably\n\n"
"-P perform peak tracking\n\n"
"-fmin <float> miniumum F0 value. Sets the minimum allowed F0 in \n"
" output track. Default is "+ftoString(DEFAULT_MIN_PITCH)+".\n "
" Changing this to suit the speaker usually increases \n"
" performance. Typical recommended values are 60-90Hz for\n"
" males and 120-150Hz for females\n\n"
"-fmax <float> maxiumum F0 value. Sets the maximum allowed F0 in \n"
" output track. Default is "+ftoString(DEFAULT_MAX_PITCH)+". \n"
" Changing this to suit the speaker usually increases \n"
" performance. Typical recommended values are 200Hz for \n"
" males and 300-400Hz for females\n\n"
"-shift <float> frame spacing in seconds for fixed frame analysis. \n"
" This doesn't have to be the same as the output file spacing - \n"
" the -S option can be used to resample the track before saving \n"
" default: "+ftoString(DEFAULT_SHIFT/1000.0) +"\n\n"
"-length <float> analysis frame length in seconds.\n"
" default: "+ftoString(DEFAULT_LENGTH/1000.0) +"\n\n"
"-lpfilter <int> Low pass filter, with cutoff frequency in Hz \n"
" Filtering is performed by a FIR filter which is built at run \n"
" time. The order of the filter can be given by -forder. The \n"
" default value is 199\n\n"
"-forder <int> Order of FIR filter used for lpfilter and \n"
" hpfilter. This must be ODD. Sensible values range \n"
" from 19 (quick but with a shallow rolloff) to 199 \n"
" (slow but with a steep rolloff). The default is 199.\n\n";
}
EST_String options_pda_srpd(void)
{
// The standard waveform input options
return
EST_String("")+
"-d <float> decimation factor\n"
" set down-sampling for quicker computation so that only one in \n"
" <parameter>decimation factor</parameter> samples are used in the first instance. \n"
" Must be in the range of one to ten inclusive. Default is four. \n"
" For data sampled at 10kHz, it is advised that a decimation \n"
" factor of two isselected.\n\n"
"-n <float> Inoise floor.\n"
" Set the maximum absolute signal amplitude that represents \n"
" silence to <parameter>Inoise floor</parameter>. If the absolute amplitude of \n"
" the first segment in a given frame is below this level at all \n"
" times, then the frame is classified as representing silence. \n"
" Must be a positive number. Default is 120 ADC units.\n\n"
"-H <float> unvoiced to voiced coeff threshold\n"
" set the correlation coefficient threshold which must be \n"
" exceeded in a transition from an unvoiced classified frame \n"
" of speech to a voiced frame as the unvoiced to voiced coeff \n"
" threshold. Must be in the range zero to one inclusive. \n"
" Default is 0.88.\n\n"
"-m <float> min voiced to unvoiced coeff threshold \n"
" set the minimum allowed correlation coefficient threshold \n"
" which must not be exceeded in a transition from a voiced \n"
" classified frame of speech to an unvoiced frame, as \n"
" <parameter>min voiced to unvoiced coeff threshold</parameter>. Must be in the \n"
" range zero to <parameter>unvoiced to voiced coeff threshold</parameter> \n"
" inclusive. Default is 0.75.\n\n"
"-R <float> voiced to unvoiced coeff threshold-ratio \n"
" set the scaling factor used in determining the correlation\n"
" coefficient threshold which must not be exceeded in a voiced \n"
" frame to unvoiced frame transition, as <parameter>voiced to unvoiced</parameter> \n"
" coeff threshold -ratio. The voiced to unvoiced coefficient \n"
" threshold is determined by multiplying this scaling factor \n"
" with the maximum cross-correlation coefficient of the \n"
" previously voiced frame. If this product is less than \n"
" <parameter>min voiced to unvoiced coeff threshold</parameter> then this is used \n"
" instead. Must be in the range zero to one inclusive. \n"
" Default is 0.85.\n\n"
"-t <float> anti pitch doubling/halving threshold\n"
" set the threshold used in eliminating (as far as possible) \n"
" pitch doubling and pitch halving errors as <parameter>anti pitch \n"
" double/halving threshold</parameter>. Must be in the range zero to \n"
" one inclusive. Default is 0.77.\n\n";
}
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