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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 <math.h>
39 #include "asterisk/plc.h"
41 #if !defined(FALSE)
42 #define FALSE 0
43 #endif
44 #if !defined(TRUE)
45 #define TRUE (!FALSE)
46 #endif
48 #if !defined(INT16_MAX)
49 #define INT16_MAX (32767)
50 #define INT16_MIN (-32767-1)
51 #endif
53 /* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
54 #define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */
56 #define ms_to_samples(t) (((t)*DEFAULT_SAMPLE_RATE)/1000)
58 static inline int16_t fsaturate(double damp)
60 if (damp > 32767.0)
61 return INT16_MAX;
62 if (damp < -32768.0)
63 return INT16_MIN;
64 return (int16_t) rint(damp);
67 static void save_history(plc_state_t *s, int16_t *buf, int len)
69 if (len >= PLC_HISTORY_LEN) {
70 /* Just keep the last part of the new data, starting at the beginning of the buffer */
71 memcpy(s->history, buf + len - PLC_HISTORY_LEN, sizeof(int16_t) * PLC_HISTORY_LEN);
72 s->buf_ptr = 0;
73 return;
75 if (s->buf_ptr + len > PLC_HISTORY_LEN) {
76 /* Wraps around - must break into two sections */
77 memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr));
78 len -= (PLC_HISTORY_LEN - s->buf_ptr);
79 memcpy(s->history, buf + (PLC_HISTORY_LEN - s->buf_ptr), sizeof(int16_t)*len);
80 s->buf_ptr = len;
81 return;
83 /* Can use just one section */
84 memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t)*len);
85 s->buf_ptr += len;
88 /*- End of function --------------------------------------------------------*/
90 static void normalise_history(plc_state_t *s)
92 int16_t tmp[PLC_HISTORY_LEN];
94 if (s->buf_ptr == 0)
95 return;
96 memcpy(tmp, s->history, sizeof(int16_t)*s->buf_ptr);
97 memcpy(s->history, s->history + s->buf_ptr, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr));
98 memcpy(s->history + PLC_HISTORY_LEN - s->buf_ptr, tmp, sizeof(int16_t) * s->buf_ptr);
99 s->buf_ptr = 0;
102 /*- End of function --------------------------------------------------------*/
104 static int __inline__ amdf_pitch(int min_pitch, int max_pitch, int16_t amp[], int len)
106 int i;
107 int j;
108 int acc;
109 int min_acc;
110 int pitch;
112 pitch = min_pitch;
113 min_acc = INT_MAX;
114 for (i = max_pitch; i <= min_pitch; i++) {
115 acc = 0;
116 for (j = 0; j < len; j++)
117 acc += abs(amp[i + j] - amp[j]);
118 if (acc < min_acc) {
119 min_acc = acc;
120 pitch = i;
123 return pitch;
126 /*- End of function --------------------------------------------------------*/
128 int plc_rx(plc_state_t *s, int16_t amp[], int len)
130 int i;
131 int pitch_overlap;
132 float old_step;
133 float new_step;
134 float old_weight;
135 float new_weight;
136 float gain;
138 if (s->missing_samples) {
139 /* Although we have a real signal, we need to smooth it to fit well
140 with the synthetic signal we used for the previous block */
142 /* The start of the real data is overlapped with the next 1/4 cycle
143 of the synthetic data. */
144 pitch_overlap = s->pitch >> 2;
145 if (pitch_overlap > len)
146 pitch_overlap = len;
147 gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
148 if (gain < 0.0)
149 gain = 0.0;
150 new_step = 1.0/pitch_overlap;
151 old_step = new_step*gain;
152 new_weight = new_step;
153 old_weight = (1.0 - new_step)*gain;
154 for (i = 0; i < pitch_overlap; i++) {
155 amp[i] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[i]);
156 if (++s->pitch_offset >= s->pitch)
157 s->pitch_offset = 0;
158 new_weight += new_step;
159 old_weight -= old_step;
160 if (old_weight < 0.0)
161 old_weight = 0.0;
163 s->missing_samples = 0;
165 save_history(s, amp, len);
166 return len;
169 /*- End of function --------------------------------------------------------*/
171 int plc_fillin(plc_state_t *s, int16_t amp[], int len)
173 int i;
174 int pitch_overlap;
175 float old_step;
176 float new_step;
177 float old_weight;
178 float new_weight;
179 float gain;
180 int16_t *orig_amp;
181 int orig_len;
183 orig_amp = amp;
184 orig_len = len;
185 if (s->missing_samples == 0) {
186 /* As the gap in real speech starts we need to assess the last known pitch,
187 and prepare the synthetic data we will use for fill-in */
188 normalise_history(s);
189 s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
190 /* We overlap a 1/4 wavelength */
191 pitch_overlap = s->pitch >> 2;
192 /* Cook up a single cycle of pitch, using a single of the real signal with 1/4
193 cycle OLA'ed to make the ends join up nicely */
194 /* The first 3/4 of the cycle is a simple copy */
195 for (i = 0; i < s->pitch - pitch_overlap; i++)
196 s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
197 /* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
198 new_step = 1.0/pitch_overlap;
199 new_weight = new_step;
200 for ( ; i < s->pitch; i++) {
201 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;
202 new_weight += new_step;
204 /* We should now be ready to fill in the gap with repeated, decaying cycles
205 of what is in pitchbuf */
207 /* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
208 it into the previous real data. To avoid the need to introduce a delay
209 in the stream, reverse the last 1/4 wavelength, and OLA with that. */
210 gain = 1.0;
211 new_step = 1.0 / pitch_overlap;
212 old_step = new_step;
213 new_weight = new_step;
214 old_weight = 1.0 - new_step;
215 for (i = 0; i < pitch_overlap; i++) {
216 amp[i] = fsaturate(old_weight * s->history[PLC_HISTORY_LEN - 1 - i] + new_weight * s->pitchbuf[i]);
217 new_weight += new_step;
218 old_weight -= old_step;
219 if (old_weight < 0.0)
220 old_weight = 0.0;
222 s->pitch_offset = i;
223 } else {
224 gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
225 i = 0;
227 for ( ; gain > 0.0 && i < len; i++) {
228 amp[i] = s->pitchbuf[s->pitch_offset] * gain;
229 gain -= ATTENUATION_INCREMENT;
230 if (++s->pitch_offset >= s->pitch)
231 s->pitch_offset = 0;
233 for ( ; i < len; i++)
234 amp[i] = 0;
235 s->missing_samples += orig_len;
236 save_history(s, amp, len);
237 return len;
240 /*- End of function --------------------------------------------------------*/
242 plc_state_t *plc_init(plc_state_t *s)
244 memset(s, 0, sizeof(*s));
245 return s;
247 /*- End of function --------------------------------------------------------*/
248 /*- End of file ------------------------------------------------------------*/