Sat Aug 6 00:40:02 2011

Asterisk developer's documentation


plc.h File Reference

SpanDSP - a series of DSP components for telephony. More...

#include <stdint.h>

Go to the source code of this file.

Data Structures

struct  plc_state_t

Defines

#define _SPANDSP_PLC_H_
#define CORRELATION_SPAN   160
#define PLC_HISTORY_LEN   (CORRELATION_SPAN + PLC_PITCH_MIN)
#define PLC_PITCH_MAX   40
#define PLC_PITCH_MIN   120
#define PLC_PITCH_OVERLAP_MAX   (PLC_PITCH_MIN >> 2)

Functions

int plc_fillin (plc_state_t *s, int16_t amp[], int len)
 Fill-in a block of missing audio samples.
plc_state_tplc_init (plc_state_t *s)
 Process a block of received V.29 modem audio samples.
int plc_rx (plc_state_t *s, int16_t amp[], int len)
 Process a block of received audio samples.


Detailed Description

SpanDSP - a series of DSP components for telephony.

plc.h

Author:
Steve Underwood <steveu@coppice.org>
Copyright (C) 2004 Steve Underwood

All rights reserved.

This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version.

This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details.

You should have received a copy of the GNU General Public License along with this program; if not, write to the Free Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.

This version may be optionally licenced under the GNU LGPL licence.

A license has been granted to Digium (via disclaimer) for the use of this code.

Definition in file plc.h.


Define Documentation

#define _SPANDSP_PLC_H_

Definition at line 34 of file plc.h.

#define CORRELATION_SPAN   160

The length over which the AMDF function looks for similarity (20 ms)

Definition at line 107 of file plc.h.

Referenced by plc_fillin().

#define PLC_HISTORY_LEN   (CORRELATION_SPAN + PLC_PITCH_MIN)

History buffer length. The buffer much also be at leat 1.25 times PLC_PITCH_MIN, but that is much smaller than the buffer needs to be for the pitch assessment.

Definition at line 111 of file plc.h.

Referenced by normalise_history(), plc_fillin(), and save_history().

#define PLC_PITCH_MAX   40

Maximum allowed pitch (200 Hz)

Definition at line 103 of file plc.h.

Referenced by plc_fillin().

#define PLC_PITCH_MIN   120

Minimum allowed pitch (66 Hz)

Definition at line 101 of file plc.h.

Referenced by plc_fillin().

#define PLC_PITCH_OVERLAP_MAX   (PLC_PITCH_MIN >> 2)

Maximum pitch OLA window

Definition at line 105 of file plc.h.


Function Documentation

int plc_fillin ( plc_state_t s,
int16_t  amp[],
int  len 
)

Fill-in a block of missing audio samples.

Fill-in a block of missing audio samples.

Parameters:
s The packet loss concealer context.
amp The audio sample buffer.
len The number of samples to be synthesised.
Returns:
The number of samples synthesized.

Definition at line 174 of file plc.c.

References amdf_pitch(), ATTENUATION_INCREMENT, CORRELATION_SPAN, fsaturate(), normalise_history(), PLC_HISTORY_LEN, PLC_PITCH_MAX, PLC_PITCH_MIN, s, and save_history().

Referenced by adjust_frame_for_plc().

00175 {
00176    int i;
00177    int pitch_overlap;
00178    float old_step;
00179    float new_step;
00180    float old_weight;
00181    float new_weight;
00182    float gain;
00183    int16_t *orig_amp;
00184    int orig_len;
00185 
00186    orig_amp = amp;
00187    orig_len = len;
00188    if (s->missing_samples == 0) {
00189       /* As the gap in real speech starts we need to assess the last known pitch,
00190          and prepare the synthetic data we will use for fill-in */
00191       normalise_history(s);
00192       s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
00193       /* We overlap a 1/4 wavelength */
00194       pitch_overlap = s->pitch >> 2;
00195       /* Cook up a single cycle of pitch, using a single of the real signal with 1/4
00196          cycle OLA'ed to make the ends join up nicely */
00197       /* The first 3/4 of the cycle is a simple copy */
00198       for (i = 0;  i < s->pitch - pitch_overlap;  i++)
00199          s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
00200       /* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
00201       new_step = 1.0/pitch_overlap;
00202       new_weight = new_step;
00203       for ( ; i < s->pitch; i++) {
00204          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;
00205          new_weight += new_step;
00206       }
00207       /* We should now be ready to fill in the gap with repeated, decaying cycles
00208          of what is in pitchbuf */
00209 
00210       /* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
00211          it into the previous real data. To avoid the need to introduce a delay
00212          in the stream, reverse the last 1/4 wavelength, and OLA with that. */
00213       gain = 1.0;
00214       new_step = 1.0 / pitch_overlap;
00215       old_step = new_step;
00216       new_weight = new_step;
00217       old_weight = 1.0 - new_step;
00218       for (i = 0; i < pitch_overlap; i++) {
00219          amp[i] = fsaturate(old_weight * s->history[PLC_HISTORY_LEN - 1 - i] + new_weight * s->pitchbuf[i]);
00220          new_weight += new_step;
00221          old_weight -= old_step;
00222          if (old_weight < 0.0)
00223             old_weight = 0.0;
00224       }
00225       s->pitch_offset = i;
00226    } else {
00227       gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
00228       i = 0;
00229    }
00230    for ( ; gain > 0.0 && i < len; i++) {
00231       amp[i] = s->pitchbuf[s->pitch_offset] * gain;
00232       gain -= ATTENUATION_INCREMENT;
00233       if (++s->pitch_offset >= s->pitch)
00234          s->pitch_offset = 0;
00235    }
00236    for ( ; i < len; i++)
00237       amp[i] = 0;
00238    s->missing_samples += orig_len;
00239    save_history(s, amp, len);
00240    return len;
00241 }

plc_state_t* plc_init ( plc_state_t s  ) 

Process a block of received V.29 modem audio samples.

Process a block of received V.29 modem audio samples.

Parameters:
s The packet loss concealer context.
Returns:
A pointer to the he packet loss concealer context.

Definition at line 245 of file plc.c.

References s.

00246 {
00247    memset(s, 0, sizeof(*s));
00248    return s;
00249 }

int plc_rx ( plc_state_t s,
int16_t  amp[],
int  len 
)

Process a block of received audio samples.

Process a block of received audio samples.

Parameters:
s The packet loss concealer context.
amp The audio sample buffer.
len The number of samples in the buffer.
Returns:
The number of samples in the buffer.

Definition at line 131 of file plc.c.

References ATTENUATION_INCREMENT, fsaturate(), s, and save_history().

Referenced by adjust_frame_for_plc().

00132 {
00133    int i;
00134    int pitch_overlap;
00135    float old_step;
00136    float new_step;
00137    float old_weight;
00138    float new_weight;
00139    float gain;
00140    
00141    if (s->missing_samples) {
00142       /* Although we have a real signal, we need to smooth it to fit well
00143       with the synthetic signal we used for the previous block */
00144 
00145       /* The start of the real data is overlapped with the next 1/4 cycle
00146          of the synthetic data. */
00147       pitch_overlap = s->pitch >> 2;
00148       if (pitch_overlap > len)
00149          pitch_overlap = len;
00150       gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
00151       if (gain < 0.0)
00152          gain = 0.0;
00153       new_step = 1.0/pitch_overlap;
00154       old_step = new_step*gain;
00155       new_weight = new_step;
00156       old_weight = (1.0 - new_step)*gain;
00157       for (i = 0; i < pitch_overlap; i++) {
00158          amp[i] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[i]);
00159          if (++s->pitch_offset >= s->pitch)
00160             s->pitch_offset = 0;
00161          new_weight += new_step;
00162          old_weight -= old_step;
00163          if (old_weight < 0.0)
00164             old_weight = 0.0;
00165       }
00166       s->missing_samples = 0;
00167    }
00168    save_history(s, amp, len);
00169    return len;
00170 }


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