SpanDSP - a series of DSP components for telephony. More...
#include "asterisk.h"
#include <math.h>
#include "asterisk/plc.h"
Go to the source code of this file.
Defines | |
#define | ATTENUATION_INCREMENT 0.0025 |
#define | FALSE 0 |
#define | INT16_MAX (32767) |
#define | INT16_MIN (-32767-1) |
#define | ms_to_samples(t) (((t)*DEFAULT_SAMPLE_RATE)/1000) |
#define | TRUE (!FALSE) |
Functions | |
static int __inline__ | amdf_pitch (int min_pitch, int max_pitch, int16_t amp[], int len) |
static int16_t | fsaturate (double damp) |
static void | normalise_history (plc_state_t *s) |
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. | |
static void | save_history (plc_state_t *s, int16_t *buf, int len) |
SpanDSP - a series of DSP components for telephony.
Definition in file plc.c.
#define ATTENUATION_INCREMENT 0.0025 |
Definition at line 54 of file plc.c.
Referenced by plc_fillin(), and plc_rx().
#define INT16_MAX (32767) |
Definition at line 49 of file plc.c.
Referenced by fsaturate().
#define INT16_MIN (-32767-1) |
Definition at line 50 of file plc.c.
Referenced by fsaturate().
static int __inline__ amdf_pitch | ( | int | min_pitch, |
int | max_pitch, | ||
int16_t | amp[], | ||
int | len | ||
) | [static] |
Definition at line 104 of file plc.c.
References len().
Referenced by plc_fillin().
{ int i; int j; int acc; int min_acc; int pitch; pitch = min_pitch; min_acc = INT_MAX; for (i = max_pitch; i <= min_pitch; i++) { acc = 0; for (j = 0; j < len; j++) acc += abs(amp[i + j] - amp[j]); if (acc < min_acc) { min_acc = acc; pitch = i; } } return pitch; }
static int16_t fsaturate | ( | double | damp | ) | [inline, static] |
static void normalise_history | ( | plc_state_t * | s | ) | [static] |
Definition at line 90 of file plc.c.
References plc_state_t::buf_ptr, plc_state_t::history, and PLC_HISTORY_LEN.
Referenced by plc_fillin().
{ int16_t tmp[PLC_HISTORY_LEN]; if (s->buf_ptr == 0) return; memcpy(tmp, s->history, sizeof(int16_t)*s->buf_ptr); memcpy(s->history, s->history + s->buf_ptr, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr)); memcpy(s->history + PLC_HISTORY_LEN - s->buf_ptr, tmp, sizeof(int16_t) * s->buf_ptr); s->buf_ptr = 0; }
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; }
static void save_history | ( | plc_state_t * | s, |
int16_t * | buf, | ||
int | len | ||
) | [static] |
Definition at line 67 of file plc.c.
References plc_state_t::buf_ptr, plc_state_t::history, len(), and PLC_HISTORY_LEN.
Referenced by plc_fillin(), and plc_rx().
{ if (len >= PLC_HISTORY_LEN) { /* Just keep the last part of the new data, starting at the beginning of the buffer */ memcpy(s->history, buf + len - PLC_HISTORY_LEN, sizeof(int16_t) * PLC_HISTORY_LEN); s->buf_ptr = 0; return; } if (s->buf_ptr + len > PLC_HISTORY_LEN) { /* Wraps around - must break into two sections */ memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr)); len -= (PLC_HISTORY_LEN - s->buf_ptr); memcpy(s->history, buf + (PLC_HISTORY_LEN - s->buf_ptr), sizeof(int16_t)*len); s->buf_ptr = len; return; } /* Can use just one section */ memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t)*len); s->buf_ptr += len; }