Mon Mar 19 11:30:52 2012

Asterisk developer's documentation


plc.h File Reference

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

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Data Structures

struct  plc_state_t

Defines

#define _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 _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 99 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 103 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 95 of file plc.h.

Referenced by plc_fillin().

#define PLC_PITCH_MIN   120

Minimum allowed pitch (66 Hz)

Definition at line 93 of file plc.h.

Referenced by plc_fillin().

#define PLC_PITCH_OVERLAP_MAX   (PLC_PITCH_MIN >> 2)

Maximum pitch OLA window

Definition at line 97 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 171 of file plc.c.

References amdf_pitch(), ATTENUATION_INCREMENT, CORRELATION_SPAN, fsaturate(), plc_state_t::history, plc_state_t::missing_samples, normalise_history(), plc_state_t::pitch, plc_state_t::pitch_offset, plc_state_t::pitchbuf, PLC_HISTORY_LEN, PLC_PITCH_MAX, PLC_PITCH_MIN, and save_history().

Referenced by adjust_frame_for_plc().

00172 {
00173    int i;
00174    int pitch_overlap;
00175    float old_step;
00176    float new_step;
00177    float old_weight;
00178    float new_weight;
00179    float gain;
00180    int orig_len;
00181 
00182    orig_len = len;
00183    if (s->missing_samples == 0) {
00184       /* As the gap in real speech starts we need to assess the last known pitch,
00185          and prepare the synthetic data we will use for fill-in */
00186       normalise_history(s);
00187       s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
00188       /* We overlap a 1/4 wavelength */
00189       pitch_overlap = s->pitch >> 2;
00190       /* Cook up a single cycle of pitch, using a single of the real signal with 1/4
00191          cycle OLA'ed to make the ends join up nicely */
00192       /* The first 3/4 of the cycle is a simple copy */
00193       for (i = 0;  i < s->pitch - pitch_overlap;  i++)
00194          s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
00195       /* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
00196       new_step = 1.0/pitch_overlap;
00197       new_weight = new_step;
00198       for ( ; i < s->pitch; i++) {
00199          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;
00200          new_weight += new_step;
00201       }
00202       /* We should now be ready to fill in the gap with repeated, decaying cycles
00203          of what is in pitchbuf */
00204 
00205       /* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
00206          it into the previous real data. To avoid the need to introduce a delay
00207          in the stream, reverse the last 1/4 wavelength, and OLA with that. */
00208       gain = 1.0;
00209       new_step = 1.0 / pitch_overlap;
00210       old_step = new_step;
00211       new_weight = new_step;
00212       old_weight = 1.0 - new_step;
00213       for (i = 0; i < pitch_overlap; i++) {
00214          amp[i] = fsaturate(old_weight * s->history[PLC_HISTORY_LEN - 1 - i] + new_weight * s->pitchbuf[i]);
00215          new_weight += new_step;
00216          old_weight -= old_step;
00217          if (old_weight < 0.0)
00218             old_weight = 0.0;
00219       }
00220       s->pitch_offset = i;
00221    } else {
00222       gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
00223       i = 0;
00224    }
00225    for ( ; gain > 0.0 && i < len; i++) {
00226       amp[i] = s->pitchbuf[s->pitch_offset] * gain;
00227       gain -= ATTENUATION_INCREMENT;
00228       if (++s->pitch_offset >= s->pitch)
00229          s->pitch_offset = 0;
00230    }
00231    for ( ; i < len; i++)
00232       amp[i] = 0;
00233    s->missing_samples += orig_len;
00234    save_history(s, amp, len);
00235    return len;
00236 }

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 240 of file plc.c.

00241 {
00242    memset(s, 0, sizeof(*s));
00243    return s;
00244 }

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 128 of file plc.c.

References ATTENUATION_INCREMENT, fsaturate(), plc_state_t::missing_samples, plc_state_t::pitch, plc_state_t::pitch_offset, plc_state_t::pitchbuf, and save_history().

Referenced by adjust_frame_for_plc().

00129 {
00130    int i;
00131    int pitch_overlap;
00132    float old_step;
00133    float new_step;
00134    float old_weight;
00135    float new_weight;
00136    float gain;
00137    
00138    if (s->missing_samples) {
00139       /* Although we have a real signal, we need to smooth it to fit well
00140       with the synthetic signal we used for the previous block */
00141 
00142       /* The start of the real data is overlapped with the next 1/4 cycle
00143          of the synthetic data. */
00144       pitch_overlap = s->pitch >> 2;
00145       if (pitch_overlap > len)
00146          pitch_overlap = len;
00147       gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
00148       if (gain < 0.0)
00149          gain = 0.0;
00150       new_step = 1.0/pitch_overlap;
00151       old_step = new_step*gain;
00152       new_weight = new_step;
00153       old_weight = (1.0 - new_step)*gain;
00154       for (i = 0; i < pitch_overlap; i++) {
00155          amp[i] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[i]);
00156          if (++s->pitch_offset >= s->pitch)
00157             s->pitch_offset = 0;
00158          new_weight += new_step;
00159          old_weight -= old_step;
00160          if (old_weight < 0.0)
00161             old_weight = 0.0;
00162       }
00163       s->missing_samples = 0;
00164    }
00165    save_history(s, amp, len);
00166    return len;
00167 }


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