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asterisk/plc.c
Kevin P. Fleming 2b8338cb52 more file version tags 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-06-06 22:12:19 +00:00

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C
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/*
* SpanDSP - a series of DSP components for telephony
*
* plc.c
*
* Written by Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2004 Steve Underwood
*
* All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* This version may be optionally licenced under the GNU LGPL licence.
* This version is disclaimed to DIGIUM for inclusion in the Asterisk project.
*/
/*! \file */
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <limits.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/plc.h"
#if !defined(FALSE)
#define FALSE 0
#endif
#if !defined(TRUE)
#define TRUE (!FALSE)
#endif
#if !defined(INT16_MAX)
#define INT16_MAX (32767)
#define INT16_MIN (-32767-1)
#endif
/* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
#define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */
#define ms_to_samples(t) (((t)*SAMPLE_RATE)/1000)
static inline int16_t fsaturate(double damp)
{
if (damp > 32767.0)
return INT16_MAX;
if (damp < -32768.0)
return INT16_MIN;
return (int16_t) rint(damp);
}
static void save_history(plc_state_t *s, int16_t *buf, int len)
{
if (len >= PLC_HISTORY_LEN)
{
/* Just keep the last part of the new data, starting at the beginning of the buffer */
memcpy(s->history, buf + len - PLC_HISTORY_LEN, sizeof(int16_t)*PLC_HISTORY_LEN);
s->buf_ptr = 0;
return;
}
if (s->buf_ptr + len > PLC_HISTORY_LEN)
{
/* Wraps around - must break into two sections */
memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t)*(PLC_HISTORY_LEN - s->buf_ptr));
len -= (PLC_HISTORY_LEN - s->buf_ptr);
memcpy(s->history, buf + (PLC_HISTORY_LEN - s->buf_ptr), sizeof(int16_t)*len);
s->buf_ptr = len;
return;
}
/* Can use just one section */
memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t)*len);
s->buf_ptr += len;
}
/*- End of function --------------------------------------------------------*/
static void normalise_history(plc_state_t *s)
{
int16_t tmp[PLC_HISTORY_LEN];
if (s->buf_ptr == 0)
return;
memcpy(tmp, s->history, sizeof(int16_t)*s->buf_ptr);
memcpy(s->history, s->history + s->buf_ptr, sizeof(int16_t)*(PLC_HISTORY_LEN - s->buf_ptr));
memcpy(s->history + PLC_HISTORY_LEN - s->buf_ptr, tmp, sizeof(int16_t)*s->buf_ptr);
s->buf_ptr = 0;
}
/*- End of function --------------------------------------------------------*/
static int __inline__ amdf_pitch(int min_pitch, int max_pitch, int16_t amp[], int len)
{
int i;
int j;
int acc;
int min_acc;
int pitch;
pitch = min_pitch;
min_acc = INT_MAX;
for (i = max_pitch; i <= min_pitch; i++)
{
acc = 0;
for (j = 0; j < len; j++)
acc += abs(amp[i + j] - amp[j]);
if (acc < min_acc)
{
min_acc = acc;
pitch = i;
}
}
return pitch;
}
/*- End of function --------------------------------------------------------*/
int plc_rx(plc_state_t *s, int16_t amp[], int len)
{
int i;
int pitch_overlap;
float old_step;
float new_step;
float old_weight;
float new_weight;
float gain;
if (s->missing_samples)
{
/* Although we have a real signal, we need to smooth it to fit well
with the synthetic signal we used for the previous block */
/* The start of the real data is overlapped with the next 1/4 cycle
of the synthetic data. */
pitch_overlap = s->pitch >> 2;
if (pitch_overlap > len)
pitch_overlap = len;
gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
if (gain < 0.0)
gain = 0.0;
new_step = 1.0/pitch_overlap;
old_step = new_step*gain;
new_weight = new_step;
old_weight = (1.0 - new_step)*gain;
for (i = 0; i < pitch_overlap; i++)
{
amp[i] = fsaturate(old_weight*s->pitchbuf[s->pitch_offset] + new_weight*amp[i]);
if (++s->pitch_offset >= s->pitch)
s->pitch_offset = 0;
new_weight += new_step;
old_weight -= old_step;
if (old_weight < 0.0)
old_weight = 0.0;
}
s->missing_samples = 0;
}
save_history(s, amp, len);
return len;
}
/*- End of function --------------------------------------------------------*/
int plc_fillin(plc_state_t *s, int16_t amp[], int len)
{
int i;
int pitch_overlap;
float old_step;
float new_step;
float old_weight;
float new_weight;
float gain;
int16_t *orig_amp;
int orig_len;
orig_amp = amp;
orig_len = len;
if (s->missing_samples == 0)
{
/* As the gap in real speech starts we need to assess the last known pitch,
and prepare the synthetic data we will use for fill-in */
normalise_history(s);
s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
/* We overlap a 1/4 wavelength */
pitch_overlap = s->pitch >> 2;
/* Cook up a single cycle of pitch, using a single of the real signal with 1/4
cycle OLA'ed to make the ends join up nicely */
/* The first 3/4 of the cycle is a simple copy */
for (i = 0; i < s->pitch - pitch_overlap; i++)
s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
/* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
new_step = 1.0/pitch_overlap;
new_weight = new_step;
for ( ; i < s->pitch; i++)
{
s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i]*(1.0 - new_weight) + s->history[PLC_HISTORY_LEN - 2*s->pitch + i]*new_weight;
new_weight += new_step;
}
/* We should now be ready to fill in the gap with repeated, decaying cycles
of what is in pitchbuf */
/* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
it into the previous real data. To avoid the need to introduce a delay
in the stream, reverse the last 1/4 wavelength, and OLA with that. */
gain = 1.0;
new_step = 1.0/pitch_overlap;
old_step = new_step;
new_weight = new_step;
old_weight = 1.0 - new_step;
for (i = 0; i < pitch_overlap; i++)
{
amp[i] = fsaturate(old_weight*s->history[PLC_HISTORY_LEN - 1 - i] + new_weight*s->pitchbuf[i]);
new_weight += new_step;
old_weight -= old_step;
if (old_weight < 0.0)
old_weight = 0.0;
}
s->pitch_offset = i;
}
else
{
gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
i = 0;
}
for ( ; gain > 0.0 && i < len; i++)
{
amp[i] = s->pitchbuf[s->pitch_offset]*gain;
gain -= ATTENUATION_INCREMENT;
if (++s->pitch_offset >= s->pitch)
s->pitch_offset = 0;
}
for ( ; i < len; i++)
amp[i] = 0;
s->missing_samples += orig_len;
save_history(s, amp, len);
return len;
}
/*- End of function --------------------------------------------------------*/
plc_state_t *plc_init(plc_state_t *s)
{
memset(s, 0, sizeof(*s));
return s;
}
/*- End of function --------------------------------------------------------*/
/*- End of file ------------------------------------------------------------*/
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