Let's also include aclocal.m4
[asterisk-bristuff.git] / main / plc.c
blob336a990308ecbc8773fec54cc7ba1c56617427d1
1 /*
2 * Asterisk -- An open source telephony toolkit.
4 * Written by Steve Underwood <steveu@coppice.org>
6 * Copyright (C) 2004 Steve Underwood
8 * All rights reserved.
10 * See http://www.asterisk.org for more information about
11 * the Asterisk project. Please do not directly contact
12 * any of the maintainers of this project for assistance;
13 * the project provides a web site, mailing lists and IRC
14 * channels for your use.
16 * This program is free software, distributed under the terms of
17 * the GNU General Public License Version 2. See the LICENSE file
18 * at the top of the source tree.
20 * This version may be optionally licenced under the GNU LGPL licence.
22 * A license has been granted to Digium (via disclaimer) for the use of
23 * this code.
26 /*! \file
28 * \brief SpanDSP - a series of DSP components for telephony
30 * \author Steve Underwood <steveu@coppice.org>
33 #include "asterisk.h"
35 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
37 #include <stdio.h>
38 #include <stdlib.h>
39 #include <string.h>
40 #include <math.h>
42 #include "asterisk/plc.h"
44 #if !defined(FALSE)
45 #define FALSE 0
46 #endif
47 #if !defined(TRUE)
48 #define TRUE (!FALSE)
49 #endif
51 #if !defined(INT16_MAX)
52 #define INT16_MAX (32767)
53 #define INT16_MIN (-32767-1)
54 #endif
56 /* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
57 #define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */
59 #define ms_to_samples(t) (((t)*DEFAULT_SAMPLE_RATE)/1000)
61 static inline int16_t fsaturate(double damp)
63 if (damp > 32767.0)
64 return INT16_MAX;
65 if (damp < -32768.0)
66 return INT16_MIN;
67 return (int16_t) rint(damp);
70 static void save_history(plc_state_t *s, int16_t *buf, int len)
72 if (len >= PLC_HISTORY_LEN) {
73 /* Just keep the last part of the new data, starting at the beginning of the buffer */
74 memcpy(s->history, buf + len - PLC_HISTORY_LEN, sizeof(int16_t) * PLC_HISTORY_LEN);
75 s->buf_ptr = 0;
76 return;
78 if (s->buf_ptr + len > PLC_HISTORY_LEN) {
79 /* Wraps around - must break into two sections */
80 memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr));
81 len -= (PLC_HISTORY_LEN - s->buf_ptr);
82 memcpy(s->history, buf + (PLC_HISTORY_LEN - s->buf_ptr), sizeof(int16_t)*len);
83 s->buf_ptr = len;
84 return;
86 /* Can use just one section */
87 memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t)*len);
88 s->buf_ptr += len;
91 /*- End of function --------------------------------------------------------*/
93 static void normalise_history(plc_state_t *s)
95 int16_t tmp[PLC_HISTORY_LEN];
97 if (s->buf_ptr == 0)
98 return;
99 memcpy(tmp, s->history, sizeof(int16_t)*s->buf_ptr);
100 memcpy(s->history, s->history + s->buf_ptr, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr));
101 memcpy(s->history + PLC_HISTORY_LEN - s->buf_ptr, tmp, sizeof(int16_t) * s->buf_ptr);
102 s->buf_ptr = 0;
105 /*- End of function --------------------------------------------------------*/
107 static int __inline__ amdf_pitch(int min_pitch, int max_pitch, int16_t amp[], int len)
109 int i;
110 int j;
111 int acc;
112 int min_acc;
113 int pitch;
115 pitch = min_pitch;
116 min_acc = INT_MAX;
117 for (i = max_pitch; i <= min_pitch; i++) {
118 acc = 0;
119 for (j = 0; j < len; j++)
120 acc += abs(amp[i + j] - amp[j]);
121 if (acc < min_acc) {
122 min_acc = acc;
123 pitch = i;
126 return pitch;
129 /*- End of function --------------------------------------------------------*/
131 int plc_rx(plc_state_t *s, int16_t amp[], int len)
133 int i;
134 int pitch_overlap;
135 float old_step;
136 float new_step;
137 float old_weight;
138 float new_weight;
139 float gain;
141 if (s->missing_samples) {
142 /* Although we have a real signal, we need to smooth it to fit well
143 with the synthetic signal we used for the previous block */
145 /* The start of the real data is overlapped with the next 1/4 cycle
146 of the synthetic data. */
147 pitch_overlap = s->pitch >> 2;
148 if (pitch_overlap > len)
149 pitch_overlap = len;
150 gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
151 if (gain < 0.0)
152 gain = 0.0;
153 new_step = 1.0/pitch_overlap;
154 old_step = new_step*gain;
155 new_weight = new_step;
156 old_weight = (1.0 - new_step)*gain;
157 for (i = 0; i < pitch_overlap; i++) {
158 amp[i] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[i]);
159 if (++s->pitch_offset >= s->pitch)
160 s->pitch_offset = 0;
161 new_weight += new_step;
162 old_weight -= old_step;
163 if (old_weight < 0.0)
164 old_weight = 0.0;
166 s->missing_samples = 0;
168 save_history(s, amp, len);
169 return len;
172 /*- End of function --------------------------------------------------------*/
174 int plc_fillin(plc_state_t *s, int16_t amp[], int len)
176 int i;
177 int pitch_overlap;
178 float old_step;
179 float new_step;
180 float old_weight;
181 float new_weight;
182 float gain;
183 int16_t *orig_amp;
184 int orig_len;
186 orig_amp = amp;
187 orig_len = len;
188 if (s->missing_samples == 0) {
189 /* As the gap in real speech starts we need to assess the last known pitch,
190 and prepare the synthetic data we will use for fill-in */
191 normalise_history(s);
192 s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
193 /* We overlap a 1/4 wavelength */
194 pitch_overlap = s->pitch >> 2;
195 /* Cook up a single cycle of pitch, using a single of the real signal with 1/4
196 cycle OLA'ed to make the ends join up nicely */
197 /* The first 3/4 of the cycle is a simple copy */
198 for (i = 0; i < s->pitch - pitch_overlap; i++)
199 s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
200 /* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
201 new_step = 1.0/pitch_overlap;
202 new_weight = new_step;
203 for ( ; i < s->pitch; i++) {
204 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;
205 new_weight += new_step;
207 /* We should now be ready to fill in the gap with repeated, decaying cycles
208 of what is in pitchbuf */
210 /* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
211 it into the previous real data. To avoid the need to introduce a delay
212 in the stream, reverse the last 1/4 wavelength, and OLA with that. */
213 gain = 1.0;
214 new_step = 1.0 / pitch_overlap;
215 old_step = new_step;
216 new_weight = new_step;
217 old_weight = 1.0 - new_step;
218 for (i = 0; i < pitch_overlap; i++) {
219 amp[i] = fsaturate(old_weight * s->history[PLC_HISTORY_LEN - 1 - i] + new_weight * s->pitchbuf[i]);
220 new_weight += new_step;
221 old_weight -= old_step;
222 if (old_weight < 0.0)
223 old_weight = 0.0;
225 s->pitch_offset = i;
226 } else {
227 gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
228 i = 0;
230 for ( ; gain > 0.0 && i < len; i++) {
231 amp[i] = s->pitchbuf[s->pitch_offset] * gain;
232 gain -= ATTENUATION_INCREMENT;
233 if (++s->pitch_offset >= s->pitch)
234 s->pitch_offset = 0;
236 for ( ; i < len; i++)
237 amp[i] = 0;
238 s->missing_samples += orig_len;
239 save_history(s, amp, len);
240 return len;
243 /*- End of function --------------------------------------------------------*/
245 plc_state_t *plc_init(plc_state_t *s)
247 memset(s, 0, sizeof(*s));
248 return s;
250 /*- End of function --------------------------------------------------------*/
251 /*- End of file ------------------------------------------------------------*/