Fix directed pickup to a call that is up (bug #5425 with mods)
[asterisk/asterisk.git] / plc.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Written by Steve Underwood <steveu@coppice.org>
5  *
6  * Copyright (C) 2004 Steve Underwood
7  *
8  * All rights reserved.
9  *
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.
15  *
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.
19  *
20  * This version may be optionally licenced under the GNU LGPL licence.
21  *
22  * This version is disclaimed to DIGIUM for inclusion in the Asterisk project.
23  */
24
25 /*
26  *
27  * SpanDSP - a series of DSP components for telephony
28  *
29  */
30
31 /*! \file */
32
33 #include <stdio.h>
34 #include <stdlib.h>
35 #include <string.h>
36 #include <math.h>
37 #include <limits.h>
38
39 #include "asterisk.h"
40
41 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
42
43 #include "asterisk/plc.h"
44
45 #if !defined(FALSE)
46 #define FALSE 0
47 #endif
48 #if !defined(TRUE)
49 #define TRUE (!FALSE)
50 #endif
51
52 #if !defined(INT16_MAX)
53 #define INT16_MAX       (32767)
54 #define INT16_MIN       (-32767-1)
55 #endif
56
57 /* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
58 #define ATTENUATION_INCREMENT       0.0025                            /* Attenuation per sample */
59
60 #define ms_to_samples(t)            (((t)*SAMPLE_RATE)/1000)
61
62 static inline int16_t fsaturate(double damp)
63 {
64         if (damp > 32767.0)
65                 return  INT16_MAX;
66         if (damp < -32768.0)
67                 return  INT16_MIN;
68         return (int16_t) rint(damp);
69 }
70
71 static void save_history(plc_state_t *s, int16_t *buf, int len)
72 {
73         if (len >= PLC_HISTORY_LEN) {
74                 /* Just keep the last part of the new data, starting at the beginning of the buffer */
75                  memcpy(s->history, buf + len - PLC_HISTORY_LEN, sizeof(int16_t)*PLC_HISTORY_LEN);
76                 s->buf_ptr = 0;
77                 return;
78         }
79         if (s->buf_ptr + len > PLC_HISTORY_LEN) {
80                 /* Wraps around - must break into two sections */
81                 memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t)*(PLC_HISTORY_LEN - s->buf_ptr));
82                 len -= (PLC_HISTORY_LEN - s->buf_ptr);
83                 memcpy(s->history, buf + (PLC_HISTORY_LEN - s->buf_ptr), sizeof(int16_t)*len);
84                 s->buf_ptr = len;
85                 return;
86         }
87         /* Can use just one section */
88         memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t)*len);
89         s->buf_ptr += len;
90 }
91
92 /*- End of function --------------------------------------------------------*/
93
94 static void normalise_history(plc_state_t *s)
95 {
96         int16_t tmp[PLC_HISTORY_LEN];
97
98         if (s->buf_ptr == 0)
99                 return;
100         memcpy(tmp, s->history, sizeof(int16_t)*s->buf_ptr);
101         memcpy(s->history, s->history + s->buf_ptr, sizeof(int16_t)*(PLC_HISTORY_LEN - s->buf_ptr));
102         memcpy(s->history + PLC_HISTORY_LEN - s->buf_ptr, tmp, sizeof(int16_t)*s->buf_ptr);
103         s->buf_ptr = 0;
104 }
105
106 /*- End of function --------------------------------------------------------*/
107
108 static int __inline__ amdf_pitch(int min_pitch, int max_pitch, int16_t amp[], int len)
109 {
110         int i;
111         int j;
112         int acc;
113         int min_acc;
114         int pitch;
115
116         pitch = min_pitch;
117         min_acc = INT_MAX;
118         for (i = max_pitch;  i <= min_pitch;  i++) {
119                 acc = 0;
120                 for (j = 0;  j < len;  j++)
121                         acc += abs(amp[i + j] - amp[j]);
122                 if (acc < min_acc) {
123                         min_acc = acc;
124                         pitch = i;
125                 }
126         }
127         return pitch;
128 }
129
130 /*- End of function --------------------------------------------------------*/
131
132 int plc_rx(plc_state_t *s, int16_t amp[], int len)
133 {
134         int i;
135         int pitch_overlap;
136         float old_step;
137         float new_step;
138         float old_weight;
139         float new_weight;
140         float gain;
141         
142         if (s->missing_samples) {
143                 /* Although we have a real signal, we need to smooth it to fit well
144                 with the synthetic signal we used for the previous block */
145
146                 /* The start of the real data is overlapped with the next 1/4 cycle
147                    of the synthetic data. */
148                 pitch_overlap = s->pitch >> 2;
149                 if (pitch_overlap > len)
150                         pitch_overlap = len;
151                 gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
152                 if (gain < 0.0)
153                         gain = 0.0;
154                 new_step = 1.0/pitch_overlap;
155                 old_step = new_step*gain;
156                 new_weight = new_step;
157                 old_weight = (1.0 - new_step)*gain;
158                 for (i = 0;  i < pitch_overlap;  i++) {
159                         amp[i] = fsaturate(old_weight*s->pitchbuf[s->pitch_offset] + new_weight*amp[i]);
160                         if (++s->pitch_offset >= s->pitch)
161                                 s->pitch_offset = 0;
162                         new_weight += new_step;
163                         old_weight -= old_step;
164                         if (old_weight < 0.0)
165                                 old_weight = 0.0;
166                 }
167                 s->missing_samples = 0;
168         }
169         save_history(s, amp, len);
170         return len;
171 }
172
173 /*- End of function --------------------------------------------------------*/
174
175 int plc_fillin(plc_state_t *s, int16_t amp[], int len)
176 {
177         int i;
178         int pitch_overlap;
179         float old_step;
180         float new_step;
181         float old_weight;
182         float new_weight;
183         float gain;
184         int16_t *orig_amp;
185         int orig_len;
186
187         orig_amp = amp;
188         orig_len = len;
189         if (s->missing_samples == 0) {
190                 /* As the gap in real speech starts we need to assess the last known pitch,
191                 and prepare the synthetic data we will use for fill-in */
192                 normalise_history(s);
193                 s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
194                 /* We overlap a 1/4 wavelength */
195                 pitch_overlap = s->pitch >> 2;
196                 /* Cook up a single cycle of pitch, using a single of the real signal with 1/4
197                 cycle OLA'ed to make the ends join up nicely */
198                 /* The first 3/4 of the cycle is a simple copy */
199                 for (i = 0;  i < s->pitch - pitch_overlap;  i++)
200                         s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
201                 /* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
202                 new_step = 1.0/pitch_overlap;
203                 new_weight = new_step;
204                 for (  ;  i < s->pitch;  i++) {
205                         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;
206                         new_weight += new_step;
207                 }
208                 /* We should now be ready to fill in the gap with repeated, decaying cycles
209                 of what is in pitchbuf */
210
211                 /* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
212                 it into the previous real data. To avoid the need to introduce a delay
213                 in the stream, reverse the last 1/4 wavelength, and OLA with that. */
214                 gain = 1.0;
215                 new_step = 1.0/pitch_overlap;
216                 old_step = new_step;
217                 new_weight = new_step;
218                 old_weight = 1.0 - new_step;
219                 for (i = 0;  i < pitch_overlap;  i++) {
220                         amp[i] = fsaturate(old_weight*s->history[PLC_HISTORY_LEN - 1 - i] + new_weight*s->pitchbuf[i]);
221                         new_weight += new_step;
222                         old_weight -= old_step;
223                         if (old_weight < 0.0)
224                                 old_weight = 0.0;
225                 }
226                 s->pitch_offset = i;
227         } else {
228                 gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
229                 i = 0;
230         }
231         for (  ;  gain > 0.0  &&  i < len;  i++) {
232                 amp[i] = s->pitchbuf[s->pitch_offset]*gain;
233                 gain -= ATTENUATION_INCREMENT;
234                 if (++s->pitch_offset >= s->pitch)
235                         s->pitch_offset = 0;
236         }
237         for (  ;  i < len;  i++)
238                 amp[i] = 0;
239         s->missing_samples += orig_len;
240         save_history(s, amp, len);
241         return len;
242 }
243
244 /*- End of function --------------------------------------------------------*/
245
246 plc_state_t *plc_init(plc_state_t *s)
247 {
248         memset(s, 0, sizeof(*s));
249         return s;
250 }
251 /*- End of function --------------------------------------------------------*/
252 /*- End of file ------------------------------------------------------------*/