plc.c

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Written by Steve Underwood <steveu@coppice.org>
 *
 * Copyright (C) 2004 Steve Underwood
 *
 * All rights reserved.
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 *
 * This version may be optionally licenced under the GNU LGPL licence.
 *
 * This version is disclaimed to DIGIUM for inclusion in the Asterisk project.
 */

/*! \file
 *
 * \brief SpanDSP - a series of DSP components for telephony
 *
 * \author Steve Underwood <steveu@coppice.org>
 */

#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)*DEFAULT_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 ------------------------------------------------------------*/