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merge new_loader_completion branch, including (at least):
- restructured build tree and makefiles to eliminate recursion problems - support for embedded modules - support for static builds - simpler cross-compilation support - simpler module/loader interface (no exported symbols) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
251
main/plc.c
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251
main/plc.c
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Written by Steve Underwood <steveu@coppice.org>
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*
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* Copyright (C) 2004 Steve Underwood
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*
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* All rights reserved.
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*
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* This version may be optionally licenced under the GNU LGPL licence.
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*
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* A license has been granted to Digium (via disclaimer) for the use of
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* this code.
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*/
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/*! \file
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*
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* \brief SpanDSP - a series of DSP components for telephony
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*
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* \author Steve Underwood <steveu@coppice.org>
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*/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <math.h>
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#include "asterisk/plc.h"
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#if !defined(FALSE)
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#define FALSE 0
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#endif
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#if !defined(TRUE)
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#define TRUE (!FALSE)
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#endif
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#if !defined(INT16_MAX)
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#define INT16_MAX (32767)
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#define INT16_MIN (-32767-1)
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#endif
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/* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
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#define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */
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#define ms_to_samples(t) (((t)*DEFAULT_SAMPLE_RATE)/1000)
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static inline int16_t fsaturate(double damp)
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{
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if (damp > 32767.0)
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return INT16_MAX;
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if (damp < -32768.0)
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return INT16_MIN;
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return (int16_t) rint(damp);
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}
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static void save_history(plc_state_t *s, int16_t *buf, int len)
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{
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if (len >= PLC_HISTORY_LEN) {
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/* Just keep the last part of the new data, starting at the beginning of the buffer */
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memcpy(s->history, buf + len - PLC_HISTORY_LEN, sizeof(int16_t) * PLC_HISTORY_LEN);
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s->buf_ptr = 0;
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return;
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}
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if (s->buf_ptr + len > PLC_HISTORY_LEN) {
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/* Wraps around - must break into two sections */
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memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr));
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len -= (PLC_HISTORY_LEN - s->buf_ptr);
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memcpy(s->history, buf + (PLC_HISTORY_LEN - s->buf_ptr), sizeof(int16_t)*len);
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s->buf_ptr = len;
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return;
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}
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/* Can use just one section */
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memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t)*len);
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s->buf_ptr += len;
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}
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/*- End of function --------------------------------------------------------*/
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static void normalise_history(plc_state_t *s)
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{
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int16_t tmp[PLC_HISTORY_LEN];
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if (s->buf_ptr == 0)
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return;
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memcpy(tmp, s->history, sizeof(int16_t)*s->buf_ptr);
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memcpy(s->history, s->history + s->buf_ptr, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr));
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memcpy(s->history + PLC_HISTORY_LEN - s->buf_ptr, tmp, sizeof(int16_t) * s->buf_ptr);
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s->buf_ptr = 0;
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}
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/*- End of function --------------------------------------------------------*/
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static int __inline__ amdf_pitch(int min_pitch, int max_pitch, int16_t amp[], int len)
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{
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int i;
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int j;
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int acc;
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int min_acc;
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int pitch;
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pitch = min_pitch;
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min_acc = INT_MAX;
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for (i = max_pitch; i <= min_pitch; i++) {
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acc = 0;
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for (j = 0; j < len; j++)
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acc += abs(amp[i + j] - amp[j]);
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if (acc < min_acc) {
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min_acc = acc;
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pitch = i;
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}
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}
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return pitch;
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}
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/*- End of function --------------------------------------------------------*/
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int plc_rx(plc_state_t *s, int16_t amp[], int len)
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{
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int i;
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int pitch_overlap;
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float old_step;
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float new_step;
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float old_weight;
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float new_weight;
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float gain;
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if (s->missing_samples) {
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/* Although we have a real signal, we need to smooth it to fit well
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with the synthetic signal we used for the previous block */
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/* The start of the real data is overlapped with the next 1/4 cycle
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of the synthetic data. */
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pitch_overlap = s->pitch >> 2;
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if (pitch_overlap > len)
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pitch_overlap = len;
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gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
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if (gain < 0.0)
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gain = 0.0;
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new_step = 1.0/pitch_overlap;
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old_step = new_step*gain;
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new_weight = new_step;
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old_weight = (1.0 - new_step)*gain;
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for (i = 0; i < pitch_overlap; i++) {
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amp[i] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[i]);
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if (++s->pitch_offset >= s->pitch)
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s->pitch_offset = 0;
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new_weight += new_step;
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old_weight -= old_step;
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if (old_weight < 0.0)
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old_weight = 0.0;
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}
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s->missing_samples = 0;
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}
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save_history(s, amp, len);
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return len;
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}
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/*- End of function --------------------------------------------------------*/
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int plc_fillin(plc_state_t *s, int16_t amp[], int len)
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{
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int i;
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int pitch_overlap;
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float old_step;
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float new_step;
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float old_weight;
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float new_weight;
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float gain;
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int16_t *orig_amp;
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int orig_len;
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orig_amp = amp;
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orig_len = len;
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if (s->missing_samples == 0) {
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/* As the gap in real speech starts we need to assess the last known pitch,
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and prepare the synthetic data we will use for fill-in */
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normalise_history(s);
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s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
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/* We overlap a 1/4 wavelength */
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pitch_overlap = s->pitch >> 2;
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/* Cook up a single cycle of pitch, using a single of the real signal with 1/4
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cycle OLA'ed to make the ends join up nicely */
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/* The first 3/4 of the cycle is a simple copy */
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for (i = 0; i < s->pitch - pitch_overlap; i++)
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s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
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/* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
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new_step = 1.0/pitch_overlap;
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new_weight = new_step;
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for ( ; i < s->pitch; i++) {
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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;
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new_weight += new_step;
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}
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/* We should now be ready to fill in the gap with repeated, decaying cycles
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of what is in pitchbuf */
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/* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
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it into the previous real data. To avoid the need to introduce a delay
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in the stream, reverse the last 1/4 wavelength, and OLA with that. */
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gain = 1.0;
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new_step = 1.0 / pitch_overlap;
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old_step = new_step;
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new_weight = new_step;
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old_weight = 1.0 - new_step;
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for (i = 0; i < pitch_overlap; i++) {
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amp[i] = fsaturate(old_weight * s->history[PLC_HISTORY_LEN - 1 - i] + new_weight * s->pitchbuf[i]);
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new_weight += new_step;
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old_weight -= old_step;
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if (old_weight < 0.0)
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old_weight = 0.0;
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}
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s->pitch_offset = i;
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} else {
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gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
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i = 0;
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}
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for ( ; gain > 0.0 && i < len; i++) {
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amp[i] = s->pitchbuf[s->pitch_offset] * gain;
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gain -= ATTENUATION_INCREMENT;
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if (++s->pitch_offset >= s->pitch)
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s->pitch_offset = 0;
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}
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for ( ; i < len; i++)
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amp[i] = 0;
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s->missing_samples += orig_len;
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save_history(s, amp, len);
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return len;
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}
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/*- End of function --------------------------------------------------------*/
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plc_state_t *plc_init(plc_state_t *s)
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{
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memset(s, 0, sizeof(*s));
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return s;
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}
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/*- End of function --------------------------------------------------------*/
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/*- End of file ------------------------------------------------------------*/
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