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 171 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; int16_t *orig_amp; int orig_len; orig_amp = amp; 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 | ) |
Process a block of received V.29 modem audio samples.
Process a block of received V.29 modem audio samples.
s | The packet loss concealer context. |
Definition at line 242 of file plc.c.
References s.
{ memset(s, 0, sizeof(*s)); return 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 128 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; }