aboutsummaryrefslogtreecommitdiffstats
path: root/trunk/main/plc.c
diff options
context:
space:
mode:
Diffstat (limited to 'trunk/main/plc.c')
-rw-r--r--trunk/main/plc.c248
1 files changed, 248 insertions, 0 deletions
diff --git a/trunk/main/plc.c b/trunk/main/plc.c
new file mode 100644
index 000000000..ef549ca2c
--- /dev/null
+++ b/trunk/main/plc.c
@@ -0,0 +1,248 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2004 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ *
+ * 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.
+ */
+
+/*! \file
+ *
+ * \brief SpanDSP - a series of DSP components for telephony
+ *
+ * \author Steve Underwood <steveu@coppice.org>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <math.h>
+
+#include "asterisk/plc.h"
+
+#if !defined(FALSE)
+#define FALSE 0
+#endif
+#if !defined(TRUE)
+#define TRUE (!FALSE)
+#endif
+
+#if !defined(INT16_MAX)
+#define INT16_MAX (32767)
+#define INT16_MIN (-32767-1)
+#endif
+
+/* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
+#define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */
+
+#define ms_to_samples(t) (((t)*DEFAULT_SAMPLE_RATE)/1000)
+
+static inline int16_t fsaturate(double damp)
+{
+ if (damp > 32767.0)
+ return INT16_MAX;
+ if (damp < -32768.0)
+ return INT16_MIN;
+ return (int16_t) rint(damp);
+}
+
+static void save_history(plc_state_t *s, int16_t *buf, int len)
+{
+ 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;
+}
+
+/*- End of function --------------------------------------------------------*/
+
+static void normalise_history(plc_state_t *s)
+{
+ 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;
+}
+
+/*- End of function --------------------------------------------------------*/
+
+static int __inline__ amdf_pitch(int min_pitch, int max_pitch, int16_t amp[], int len)
+{
+ 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;
+}
+
+/*- End of function --------------------------------------------------------*/
+
+int plc_rx(plc_state_t *s, int16_t amp[], int len)
+{
+ 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;
+}
+
+/*- End of function --------------------------------------------------------*/
+
+int plc_fillin(plc_state_t *s, int16_t amp[], int len)
+{
+ 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;
+}
+
+/*- End of function --------------------------------------------------------*/
+
+plc_state_t *plc_init(plc_state_t *s)
+{
+ memset(s, 0, sizeof(*s));
+ return s;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/