Mon Oct 8 12:39:04 2012

Asterisk developer's documentation


plc.c

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00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Written by Steve Underwood <steveu@coppice.org>
00005  *
00006  * Copyright (C) 2004 Steve Underwood
00007  *
00008  * All rights reserved.
00009  *
00010  * See http://www.asterisk.org for more information about
00011  * the Asterisk project. Please do not directly contact
00012  * any of the maintainers of this project for assistance;
00013  * the project provides a web site, mailing lists and IRC
00014  * channels for your use.
00015  *
00016  * This program is free software, distributed under the terms of
00017  * the GNU General Public License Version 2. See the LICENSE file
00018  * at the top of the source tree.
00019  *
00020  * This version may be optionally licenced under the GNU LGPL licence.
00021  *
00022  * A license has been granted to Digium (via disclaimer) for the use of
00023  * this code.
00024  */
00025 
00026 /*! \file
00027  *
00028  * \brief SpanDSP - a series of DSP components for telephony
00029  *
00030  * \author Steve Underwood <steveu@coppice.org>
00031  */
00032 
00033 /*** MODULEINFO
00034    <support_level>core</support_level>
00035  ***/
00036 
00037 #include "asterisk.h"
00038 
00039 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 369001 $")
00040 
00041 #include <math.h>
00042 
00043 #include "asterisk/plc.h"
00044 
00045 #if !defined(FALSE)
00046 #define FALSE 0
00047 #endif
00048 #if !defined(TRUE)
00049 #define TRUE (!FALSE)
00050 #endif
00051 
00052 #if !defined(INT16_MAX)
00053 #define INT16_MAX (32767)
00054 #define INT16_MIN (-32767-1)
00055 #endif
00056 
00057 /* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
00058 #define ATTENUATION_INCREMENT       0.0025               /* Attenuation per sample */
00059 
00060 #define ms_to_samples(t)       (((t)*DEFAULT_SAMPLE_RATE)/1000)
00061 
00062 static inline int16_t fsaturate(double damp)
00063 {
00064    if (damp > 32767.0)
00065       return  INT16_MAX;
00066    if (damp < -32768.0)
00067       return  INT16_MIN;
00068    return (int16_t) rint(damp);
00069 }
00070 
00071 static void save_history(plc_state_t *s, int16_t *buf, int len)
00072 {
00073    if (len >= PLC_HISTORY_LEN) {
00074       /* Just keep the last part of the new data, starting at the beginning of the buffer */
00075        memcpy(s->history, buf + len - PLC_HISTORY_LEN, sizeof(int16_t) * PLC_HISTORY_LEN);
00076       s->buf_ptr = 0;
00077       return;
00078    }
00079    if (s->buf_ptr + len > PLC_HISTORY_LEN) {
00080       /* Wraps around - must break into two sections */
00081       memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr));
00082       len -= (PLC_HISTORY_LEN - s->buf_ptr);
00083       memcpy(s->history, buf + (PLC_HISTORY_LEN - s->buf_ptr), sizeof(int16_t)*len);
00084       s->buf_ptr = len;
00085       return;
00086    }
00087    /* Can use just one section */
00088    memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t)*len);
00089    s->buf_ptr += len;
00090 }
00091 
00092 /*- End of function --------------------------------------------------------*/
00093 
00094 static void normalise_history(plc_state_t *s)
00095 {
00096    int16_t tmp[PLC_HISTORY_LEN];
00097 
00098    if (s->buf_ptr == 0)
00099       return;
00100    memcpy(tmp, s->history, sizeof(int16_t)*s->buf_ptr);
00101    memcpy(s->history, s->history + s->buf_ptr, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr));
00102    memcpy(s->history + PLC_HISTORY_LEN - s->buf_ptr, tmp, sizeof(int16_t) * s->buf_ptr);
00103    s->buf_ptr = 0;
00104 }
00105 
00106 /*- End of function --------------------------------------------------------*/
00107 
00108 static int __inline__ amdf_pitch(int min_pitch, int max_pitch, int16_t amp[], int len)
00109 {
00110    int i;
00111    int j;
00112    int acc;
00113    int min_acc;
00114    int pitch;
00115 
00116    pitch = min_pitch;
00117    min_acc = INT_MAX;
00118    for (i = max_pitch; i <= min_pitch; i++) {
00119       acc = 0;
00120       for (j = 0; j < len; j++)
00121          acc += abs(amp[i + j] - amp[j]);
00122       if (acc < min_acc) {
00123          min_acc = acc;
00124          pitch = i;
00125       }
00126    }
00127    return pitch;
00128 }
00129 
00130 /*- End of function --------------------------------------------------------*/
00131 
00132 int plc_rx(plc_state_t *s, int16_t amp[], int len)
00133 {
00134    int i;
00135    int pitch_overlap;
00136    float old_step;
00137    float new_step;
00138    float old_weight;
00139    float new_weight;
00140    float gain;
00141    
00142    if (s->missing_samples) {
00143       /* Although we have a real signal, we need to smooth it to fit well
00144       with the synthetic signal we used for the previous block */
00145 
00146       /* The start of the real data is overlapped with the next 1/4 cycle
00147          of the synthetic data. */
00148       pitch_overlap = s->pitch >> 2;
00149       if (pitch_overlap > len)
00150          pitch_overlap = len;
00151       gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
00152       if (gain < 0.0)
00153          gain = 0.0;
00154       new_step = 1.0/pitch_overlap;
00155       old_step = new_step*gain;
00156       new_weight = new_step;
00157       old_weight = (1.0 - new_step)*gain;
00158       for (i = 0; i < pitch_overlap; i++) {
00159          amp[i] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[i]);
00160          if (++s->pitch_offset >= s->pitch)
00161             s->pitch_offset = 0;
00162          new_weight += new_step;
00163          old_weight -= old_step;
00164          if (old_weight < 0.0)
00165             old_weight = 0.0;
00166       }
00167       s->missing_samples = 0;
00168    }
00169    save_history(s, amp, len);
00170    return len;
00171 }
00172 
00173 /*- End of function --------------------------------------------------------*/
00174 
00175 int plc_fillin(plc_state_t *s, int16_t amp[], int len)
00176 {
00177    int i;
00178    int pitch_overlap;
00179    float old_step;
00180    float new_step;
00181    float old_weight;
00182    float new_weight;
00183    float gain;
00184    int orig_len;
00185 
00186    orig_len = len;
00187    if (s->missing_samples == 0) {
00188       /* As the gap in real speech starts we need to assess the last known pitch,
00189          and prepare the synthetic data we will use for fill-in */
00190       normalise_history(s);
00191       s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
00192       /* We overlap a 1/4 wavelength */
00193       pitch_overlap = s->pitch >> 2;
00194       /* Cook up a single cycle of pitch, using a single of the real signal with 1/4
00195          cycle OLA'ed to make the ends join up nicely */
00196       /* The first 3/4 of the cycle is a simple copy */
00197       for (i = 0;  i < s->pitch - pitch_overlap;  i++)
00198          s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
00199       /* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
00200       new_step = 1.0/pitch_overlap;
00201       new_weight = new_step;
00202       for ( ; i < s->pitch; i++) {
00203          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;
00204          new_weight += new_step;
00205       }
00206       /* We should now be ready to fill in the gap with repeated, decaying cycles
00207          of what is in pitchbuf */
00208 
00209       /* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
00210          it into the previous real data. To avoid the need to introduce a delay
00211          in the stream, reverse the last 1/4 wavelength, and OLA with that. */
00212       gain = 1.0;
00213       new_step = 1.0 / pitch_overlap;
00214       old_step = new_step;
00215       new_weight = new_step;
00216       old_weight = 1.0 - new_step;
00217       for (i = 0; i < pitch_overlap; i++) {
00218          amp[i] = fsaturate(old_weight * s->history[PLC_HISTORY_LEN - 1 - i] + new_weight * s->pitchbuf[i]);
00219          new_weight += new_step;
00220          old_weight -= old_step;
00221          if (old_weight < 0.0)
00222             old_weight = 0.0;
00223       }
00224       s->pitch_offset = i;
00225    } else {
00226       gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
00227       i = 0;
00228    }
00229    for ( ; gain > 0.0 && i < len; i++) {
00230       amp[i] = s->pitchbuf[s->pitch_offset] * gain;
00231       gain -= ATTENUATION_INCREMENT;
00232       if (++s->pitch_offset >= s->pitch)
00233          s->pitch_offset = 0;
00234    }
00235    for ( ; i < len; i++)
00236       amp[i] = 0;
00237    s->missing_samples += orig_len;
00238    save_history(s, amp, len);
00239    return len;
00240 }
00241 
00242 /*- End of function --------------------------------------------------------*/
00243 
00244 plc_state_t *plc_init(plc_state_t *s)
00245 {
00246    memset(s, 0, sizeof(*s));
00247    return s;
00248 }
00249 /*- End of function --------------------------------------------------------*/
00250 /*- End of file ------------------------------------------------------------*/

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