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-rw-r--r--1.4.23-rc4/main/plc.c251
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diff --git a/1.4.23-rc4/main/plc.c b/1.4.23-rc4/main/plc.c
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--- a/1.4.23-rc4/main/plc.c
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-/*
- * 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.
- *
- * A license has been granted to Digium (via disclaimer) for the use of
- * this code.
- */
-
-/*! \file
- *
- * \brief SpanDSP - a series of DSP components for telephony
- *
- * \author Steve Underwood <steveu@coppice.org>
- */
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-
-#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 ------------------------------------------------------------*/