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Diffstat (limited to '1.2-netsec/plc.c')
-rw-r--r-- | 1.2-netsec/plc.c | 250 |
1 files changed, 0 insertions, 250 deletions
diff --git a/1.2-netsec/plc.c b/1.2-netsec/plc.c deleted file mode 100644 index f0ad46eeb..000000000 --- a/1.2-netsec/plc.c +++ /dev/null @@ -1,250 +0,0 @@ -/* - * 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 - * - */ - -#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 ------------------------------------------------------------*/ |