This should fix issue #43, calling plc_fillin with a small value for len

actually returns more than len.

plc.c

@@ -0,0 +1,289 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * plc.c
+ *
+ * Written by Steve Underwood <[email protected]>
+ *
+ * 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 Lesser General Public License version 2.1,
+ * as published by the Free Software Foundation.
+ *
+ * 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+/*! \file */
+
+#if defined(HAVE_CONFIG_H)
+#include "config.h"
+#endif
+
+#include <stdio.h>
+#include <inttypes.h>
+#include <stdlib.h>
+#include <string.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+#include "floating_fudge.h"
+#include <limits.h>
+
+#include "spandsp/telephony.h"
+#include "spandsp/alloc.h"
+#include "spandsp/fast_convert.h"
+#include "spandsp/saturated.h"
+#include "spandsp/vector_int.h"
+#include "spandsp/plc.h"
+
+#include "spandsp/private/plc.h"
+
+/* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
+#define ATTENUATION_INCREMENT       0.0025f     /* Attenuation per sample */
+
+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 */
+        vec_copyi16(s->history, &buf[len - PLC_HISTORY_LEN], PLC_HISTORY_LEN);
+        s->buf_ptr = 0;
+        return;
+    }
+    /*endif*/
+    if (s->buf_ptr + len > PLC_HISTORY_LEN)
+    {
+        /* Wraps around - must break into two sections */
+        vec_copyi16(&s->history[s->buf_ptr], buf, PLC_HISTORY_LEN - s->buf_ptr);
+        len -= (PLC_HISTORY_LEN - s->buf_ptr);
+        vec_copyi16(s->history, &buf[PLC_HISTORY_LEN - s->buf_ptr], len);
+        s->buf_ptr = len;
+        return;
+    }
+    /*endif*/
+    /* Can use just one section */
+    vec_copyi16(&s->history[s->buf_ptr], buf, len);
+    s->buf_ptr += len;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ void normalise_history(plc_state_t *s)
+{
+    int16_t tmp[PLC_HISTORY_LEN];
+
+    if (s->buf_ptr == 0)
+        return;
+    /*endif*/
+    vec_copyi16(tmp, s->history, s->buf_ptr);
+    vec_movei16(s->history, &s->history[s->buf_ptr], PLC_HISTORY_LEN - s->buf_ptr);
+    vec_copyi16(&s->history[PLC_HISTORY_LEN - s->buf_ptr], tmp, s->buf_ptr);
+    s->buf_ptr = 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int 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]);
+        /*endfor*/
+        if (acc < min_acc)
+        {
+            min_acc = acc;
+            pitch = i;
+        }
+        /*endif*/
+    }
+    /*endfor*/
+    return pitch;
+}
+/*- End of function --------------------------------------------------------*/
+
+SPAN_DECLARE(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;
+        /*endif*/
+        gain = 1.0f - s->missing_samples*ATTENUATION_INCREMENT;
+        if (gain < 0.0f)
+            gain = 0.0f;
+        /*endif*/
+        new_step = 1.0f/pitch_overlap;
+        old_step = new_step*gain;
+        new_weight = new_step;
+        old_weight = (1.0f - 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;
+            /*endif*/
+            new_weight += new_step;
+            old_weight -= old_step;
+            if (old_weight < 0.0f)
+                old_weight = 0.0f;
+            /*endif*/
+        }
+        /*endfor*/
+        s->missing_samples = 0;
+    }
+    /*endif*/
+    save_history(s, amp, len);
+    return len;
+}
+/*- End of function --------------------------------------------------------*/
+
+SPAN_DECLARE(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;
+    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];
+        /*endfor*/
+        /* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
+        new_step = 1.0f/pitch_overlap;
+        new_weight = new_step;
+        for (  ;  i < s->pitch;  i++)
+        {
+            s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i]*(1.0f - new_weight) + s->history[PLC_HISTORY_LEN - 2*s->pitch + i]*new_weight;
+            new_weight += new_step;
+        }
+        /*endfor*/
+        /* We should now be ready to fill in the gap with repeated, decaying cycles
+           of what is in pitchbuf */
+
+        gain = 1.0f;
+        /* 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. */
+        new_step = 1.0f/pitch_overlap;
+        old_step = new_step;
+        new_weight = new_step;
+        old_weight = 1.0f - new_step;
+        /* Ensure we never copy too much to the output buffer. */
+        if (pitch_overlap > len)
+            pitch_overlap = len;
+        /*endif*/
+        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 < 0.0f)
+                old_weight = 0.0f;
+            /*endif*/
+        }
+        /*endfor*/
+        s->pitch_offset = i;
+    }
+    else
+    {
+        gain = 1.0f - s->missing_samples*ATTENUATION_INCREMENT;
+        i = 0;
+    }
+    /*endif*/
+    for (  ;  gain > 0.0f  &&  i < len;  i++)
+    {
+        amp[i] = (int16_t) (s->pitchbuf[s->pitch_offset]*gain);
+        gain -= ATTENUATION_INCREMENT;
+        if (++s->pitch_offset >= s->pitch)
+            s->pitch_offset = 0;
+        /*endif*/
+    }
+    /*endfor*/
+    for (  ;  i < len;  i++)
+        amp[i] = 0;
+    /*endif*/
+    s->missing_samples += orig_len;
+    save_history(s, amp, len);
+    return len;
+}
+/*- End of function --------------------------------------------------------*/
+
+SPAN_DECLARE(plc_state_t *) plc_init(plc_state_t *s)
+{
+    if (s == NULL)
+    {
+        if ((s = (plc_state_t *) span_alloc(sizeof(*s))) == NULL)
+            return NULL;
+        /*endif*/
+    }
+    /*endif*/
+    memset(s, 0, sizeof(*s));
+    return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+SPAN_DECLARE(int) plc_release(plc_state_t *s)
+{
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+SPAN_DECLARE(int) plc_free(plc_state_t *s)
+{
+    if (s)
+        span_free(s);
+    /*endif*/
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/