#include <stdint.h>
Go to the source code of this file.
Data Structures | |
struct | plc_state_t |
Defines | |
#define | _SPANDSP_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_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. |
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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 107 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 111 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 174 of file plc.c.
References amdf_pitch(), ATTENUATION_INCREMENT, CORRELATION_SPAN, fsaturate(), normalise_history(), PLC_HISTORY_LEN, PLC_PITCH_MAX, PLC_PITCH_MIN, s, and save_history().
Referenced by adjust_frame_for_plc().
00175 { 00176 int i; 00177 int pitch_overlap; 00178 float old_step; 00179 float new_step; 00180 float old_weight; 00181 float new_weight; 00182 float gain; 00183 int16_t *orig_amp; 00184 int orig_len; 00185 00186 orig_amp = amp; 00187 orig_len = len; 00188 if (s->missing_samples == 0) { 00189 /* As the gap in real speech starts we need to assess the last known pitch, 00190 and prepare the synthetic data we will use for fill-in */ 00191 normalise_history(s); 00192 s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN); 00193 /* We overlap a 1/4 wavelength */ 00194 pitch_overlap = s->pitch >> 2; 00195 /* Cook up a single cycle of pitch, using a single of the real signal with 1/4 00196 cycle OLA'ed to make the ends join up nicely */ 00197 /* The first 3/4 of the cycle is a simple copy */ 00198 for (i = 0; i < s->pitch - pitch_overlap; i++) 00199 s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i]; 00200 /* The last 1/4 of the cycle is overlapped with the end of the previous cycle */ 00201 new_step = 1.0/pitch_overlap; 00202 new_weight = new_step; 00203 for ( ; i < s->pitch; i++) { 00204 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; 00205 new_weight += new_step; 00206 } 00207 /* We should now be ready to fill in the gap with repeated, decaying cycles 00208 of what is in pitchbuf */ 00209 00210 /* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth 00211 it into the previous real data. To avoid the need to introduce a delay 00212 in the stream, reverse the last 1/4 wavelength, and OLA with that. */ 00213 gain = 1.0; 00214 new_step = 1.0 / pitch_overlap; 00215 old_step = new_step; 00216 new_weight = new_step; 00217 old_weight = 1.0 - new_step; 00218 for (i = 0; i < pitch_overlap; i++) { 00219 amp[i] = fsaturate(old_weight * s->history[PLC_HISTORY_LEN - 1 - i] + new_weight * s->pitchbuf[i]); 00220 new_weight += new_step; 00221 old_weight -= old_step; 00222 if (old_weight < 0.0) 00223 old_weight = 0.0; 00224 } 00225 s->pitch_offset = i; 00226 } else { 00227 gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT; 00228 i = 0; 00229 } 00230 for ( ; gain > 0.0 && i < len; i++) { 00231 amp[i] = s->pitchbuf[s->pitch_offset] * gain; 00232 gain -= ATTENUATION_INCREMENT; 00233 if (++s->pitch_offset >= s->pitch) 00234 s->pitch_offset = 0; 00235 } 00236 for ( ; i < len; i++) 00237 amp[i] = 0; 00238 s->missing_samples += orig_len; 00239 save_history(s, amp, len); 00240 return len; 00241 }
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 245 of file plc.c.
References s.
00246 { 00247 memset(s, 0, sizeof(*s)); 00248 return s; 00249 }
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 131 of file plc.c.
References ATTENUATION_INCREMENT, fsaturate(), s, and save_history().
Referenced by adjust_frame_for_plc().
00132 { 00133 int i; 00134 int pitch_overlap; 00135 float old_step; 00136 float new_step; 00137 float old_weight; 00138 float new_weight; 00139 float gain; 00140 00141 if (s->missing_samples) { 00142 /* Although we have a real signal, we need to smooth it to fit well 00143 with the synthetic signal we used for the previous block */ 00144 00145 /* The start of the real data is overlapped with the next 1/4 cycle 00146 of the synthetic data. */ 00147 pitch_overlap = s->pitch >> 2; 00148 if (pitch_overlap > len) 00149 pitch_overlap = len; 00150 gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT; 00151 if (gain < 0.0) 00152 gain = 0.0; 00153 new_step = 1.0/pitch_overlap; 00154 old_step = new_step*gain; 00155 new_weight = new_step; 00156 old_weight = (1.0 - new_step)*gain; 00157 for (i = 0; i < pitch_overlap; i++) { 00158 amp[i] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[i]); 00159 if (++s->pitch_offset >= s->pitch) 00160 s->pitch_offset = 0; 00161 new_weight += new_step; 00162 old_weight -= old_step; 00163 if (old_weight < 0.0) 00164 old_weight = 0.0; 00165 } 00166 s->missing_samples = 0; 00167 } 00168 save_history(s, amp, len); 00169 return len; 00170 }