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

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Data Structures | |
| struct | plc_state_t |
Defines | |
| #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_t * | plc_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. | |
SpanDSP - a series of DSP components for telephony.
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 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 |
| #define PLC_PITCH_MIN 120 |
| #define PLC_PITCH_OVERLAP_MAX (PLC_PITCH_MIN >> 2) |
| 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.
| s | The packet loss concealer context. |
| amp | The audio sample buffer. |
| len | The number of samples to be synthesised. |
Definition at line 175 of file plc.c.
References amdf_pitch(), ATTENUATION_INCREMENT, CORRELATION_SPAN, fsaturate(), plc_state_t::history, len(), 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().
{
int i;
int pitch_overlap;
float old_step;
float new_step;
float old_weight;
float new_weight;
float gain;
int orig_len;
orig_len = len;
if (s->missing_samples == 0) {
/* As the gap in real speech starts we need to assess the last known pitch,
and prepare the synthetic data we will use for fill-in */
normalise_history(s);
s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
/* We overlap a 1/4 wavelength */
pitch_overlap = s->pitch >> 2;
/* Cook up a single cycle of pitch, using a single of the real signal with 1/4
cycle OLA'ed to make the ends join up nicely */
/* The first 3/4 of the cycle is a simple copy */
for (i = 0; i < s->pitch - pitch_overlap; i++)
s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
/* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
new_step = 1.0/pitch_overlap;
new_weight = new_step;
for ( ; i < s->pitch; i++) {
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;
new_weight += new_step;
}
/* We should now be ready to fill in the gap with repeated, decaying cycles
of what is in pitchbuf */
/* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
it into the previous real data. To avoid the need to introduce a delay
in the stream, reverse the last 1/4 wavelength, and OLA with that. */
gain = 1.0;
new_step = 1.0 / pitch_overlap;
old_step = new_step;
new_weight = new_step;
old_weight = 1.0 - new_step;
for (i = 0; i < pitch_overlap; i++) {
amp[i] = fsaturate(old_weight * s->history[PLC_HISTORY_LEN - 1 - i] + new_weight * s->pitchbuf[i]);
new_weight += new_step;
old_weight -= old_step;
if (old_weight < 0.0)
old_weight = 0.0;
}
s->pitch_offset = i;
} else {
gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
i = 0;
}
for ( ; gain > 0.0 && i < len; i++) {
amp[i] = s->pitchbuf[s->pitch_offset] * gain;
gain -= ATTENUATION_INCREMENT;
if (++s->pitch_offset >= s->pitch)
s->pitch_offset = 0;
}
for ( ; i < len; i++)
amp[i] = 0;
s->missing_samples += orig_len;
save_history(s, amp, len);
return len;
}
| plc_state_t* plc_init | ( | plc_state_t * | s | ) |
| 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.
| s | The packet loss concealer context. |
| amp | The audio sample buffer. |
| len | The number of samples in the buffer. |
Definition at line 132 of file plc.c.
References ATTENUATION_INCREMENT, fsaturate(), len(), 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().
{
int i;
int pitch_overlap;
float old_step;
float new_step;
float old_weight;
float new_weight;
float gain;
if (s->missing_samples) {
/* Although we have a real signal, we need to smooth it to fit well
with the synthetic signal we used for the previous block */
/* The start of the real data is overlapped with the next 1/4 cycle
of the synthetic data. */
pitch_overlap = s->pitch >> 2;
if (pitch_overlap > len)
pitch_overlap = len;
gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
if (gain < 0.0)
gain = 0.0;
new_step = 1.0/pitch_overlap;
old_step = new_step*gain;
new_weight = new_step;
old_weight = (1.0 - new_step)*gain;
for (i = 0; i < pitch_overlap; i++) {
amp[i] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[i]);
if (++s->pitch_offset >= s->pitch)
s->pitch_offset = 0;
new_weight += new_step;
old_weight -= old_step;
if (old_weight < 0.0)
old_weight = 0.0;
}
s->missing_samples = 0;
}
save_history(s, amp, len);
return len;
}