SpanDSP - a series of DSP components for telephony. More...
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Defines | |
| #define | ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */ |
| #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) |
Detailed Description
SpanDSP - a series of DSP components for telephony.
Definition in file plc.c.
Define Documentation
| #define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */ |
Definition at line 58 of file plc.c.
Referenced by plc_fillin(), and plc_rx().
| #define INT16_MAX (32767) |
Definition at line 53 of file plc.c.
Referenced by fsaturate().
| #define INT16_MIN (-32767-1) |
Definition at line 54 of file plc.c.
Referenced by fsaturate().
| #define ms_to_samples | ( | t | ) | (((t)*DEFAULT_SAMPLE_RATE)/1000) |
Function Documentation
| static int __inline__ amdf_pitch | ( | int | min_pitch, |
| int | max_pitch, | ||
| int16_t | amp[], | ||
| int | len | ||
| ) | [static] |
Definition at line 108 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 94 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.
- 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 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.
- 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 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;
}
| static void save_history | ( | plc_state_t * | s, |
| int16_t * | buf, | ||
| int | len | ||
| ) | [static] |
Definition at line 71 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;
}
