Improve documentation by making all of the colors used readable,
[asterisk/asterisk.git] / main / 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  * A license has been granted to Digium (via disclaimer) for the use of
23  * this code.
24  */
25
26 /*! \file
27  *
28  * \brief SpanDSP - a series of DSP components for telephony
29  *
30  * \author Steve Underwood <steveu@coppice.org>
31  */
32
33 /*** MODULEINFO
34         <support_level>core</support_level>
35  ***/
36
37 #include "asterisk.h"
38
39 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
40
41 #include <math.h>
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)*DEFAULT_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         int orig_len;
185
186         orig_len = len;
187         if (s->missing_samples == 0) {
188                 /* As the gap in real speech starts we need to assess the last known pitch,
189                 and prepare the synthetic data we will use for fill-in */
190                 normalise_history(s);
191                 s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
192                 /* We overlap a 1/4 wavelength */
193                 pitch_overlap = s->pitch >> 2;
194                 /* Cook up a single cycle of pitch, using a single of the real signal with 1/4
195                 cycle OLA'ed to make the ends join up nicely */
196                 /* The first 3/4 of the cycle is a simple copy */
197                 for (i = 0;  i < s->pitch - pitch_overlap;  i++)
198                         s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
199                 /* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
200                 new_step = 1.0/pitch_overlap;
201                 new_weight = new_step;
202                 for ( ; i < s->pitch; i++) {
203                         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;
204                         new_weight += new_step;
205                 }
206                 /* We should now be ready to fill in the gap with repeated, decaying cycles
207                 of what is in pitchbuf */
208
209                 /* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
210                 it into the previous real data. To avoid the need to introduce a delay
211                 in the stream, reverse the last 1/4 wavelength, and OLA with that. */
212                 gain = 1.0;
213                 new_step = 1.0 / pitch_overlap;
214                 old_step = new_step;
215                 new_weight = new_step;
216                 old_weight = 1.0 - new_step;
217                 for (i = 0; i < pitch_overlap; i++) {
218                         amp[i] = fsaturate(old_weight * s->history[PLC_HISTORY_LEN - 1 - i] + new_weight * s->pitchbuf[i]);
219                         new_weight += new_step;
220                         old_weight -= old_step;
221                         if (old_weight < 0.0)
222                                 old_weight = 0.0;
223                 }
224                 s->pitch_offset = i;
225         } else {
226                 gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
227                 i = 0;
228         }
229         for ( ; gain > 0.0 && i < len; i++) {
230                 amp[i] = s->pitchbuf[s->pitch_offset] * gain;
231                 gain -= ATTENUATION_INCREMENT;
232                 if (++s->pitch_offset >= s->pitch)
233                         s->pitch_offset = 0;
234         }
235         for ( ; i < len; i++)
236                 amp[i] = 0;
237         s->missing_samples += orig_len;
238         save_history(s, amp, len);
239         return len;
240 }
241
242 /*- End of function --------------------------------------------------------*/
243
244 plc_state_t *plc_init(plc_state_t *s)
245 {
246         memset(s, 0, sizeof(*s));
247         return s;
248 }
249 /*- End of function --------------------------------------------------------*/
250 /*- End of file ------------------------------------------------------------*/