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authorBenny Prijono <bennylp@teluu.com>2007-04-07 20:45:55 +0000
committerBenny Prijono <bennylp@teluu.com>2007-04-07 20:45:55 +0000
commit4e142ea98dbc56e5efb1ad729f099f9cfefd0169 (patch)
tree44f542dd0215fb8bb2dc5c8f95d37a825741920a /third_party/plc_steveu/plc_steveu.c
parentff909fc7306b566d21558a510eaad4c85fe02412 (diff)
Moved plc, resample, and mp3 to third_party
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/split-3rd-party@1175 74dad513-b988-da41-8d7b-12977e46ad98
Diffstat (limited to 'third_party/plc_steveu/plc_steveu.c')
-rw-r--r--third_party/plc_steveu/plc_steveu.c338
1 files changed, 338 insertions, 0 deletions
diff --git a/third_party/plc_steveu/plc_steveu.c b/third_party/plc_steveu/plc_steveu.c
new file mode 100644
index 00000000..3326b748
--- /dev/null
+++ b/third_party/plc_steveu/plc_steveu.c
@@ -0,0 +1,338 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * plc.c
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * 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 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.
+ * This version is disclaimed to DIGIUM for inclusion in the Asterisk project.
+ */
+
+/*! \file */
+
+#include <pjmedia/types.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+#include <limits.h>
+
+#include "plc_steveu.h"
+
+#if !defined(FALSE)
+#define FALSE 0
+#endif
+#if !defined(TRUE)
+#define TRUE (!FALSE)
+#endif
+
+#ifndef INT16_MAX
+#define INT16_MAX (32767)
+#endif
+
+#ifndef INT16_MIN
+#define INT16_MIN (-32767-1)
+#endif
+
+//#define PJ_HAS_RINT 1
+
+
+/* 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)*SAMPLE_RATE)/1000)
+
+
+#if defined(PJ_HAS_RINT) && PJ_HAS_RINT!=0
+#define RINT(d) rint(d)
+#else
+double RINT(double d)
+{
+ double f = floor(d);
+ double c = ceil(d);
+
+ if (c-d > d-f)
+ return f;
+ else if (c-d < d-f)
+ return c;
+ else if (d >= 0) {
+ if (f/2==f)
+ return f;
+ else
+ return c;
+ } else {
+ if (c/2==c)
+ return c;
+ else
+ return f;
+ }
+}
+#endif
+
+
+PJ_INLINE(pj_int16_t) fsaturate(double damp)
+{
+ if (damp > 32767.0)
+ return INT16_MAX;
+ else if (damp < -32768.0)
+ return INT16_MIN;
+ else {
+ return (pj_int16_t) RINT(damp);
+ }
+}
+
+static void save_history(plc_state_t *s, pj_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(pj_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(pj_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(pj_int16_t)*len);
+ s->buf_ptr = len;
+ return;
+ }
+ /* Can use just one section */
+ memcpy(s->history + s->buf_ptr, buf, sizeof(pj_int16_t)*len);
+ s->buf_ptr += len;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void normalise_history(plc_state_t *s)
+{
+ pj_int16_t tmp[PLC_HISTORY_LEN];
+
+ if (s->buf_ptr == 0)
+ return;
+ memcpy(tmp, s->history, sizeof(pj_int16_t)*s->buf_ptr);
+ memcpy(s->history, s->history + s->buf_ptr, sizeof(pj_int16_t)*(PLC_HISTORY_LEN - s->buf_ptr));
+ memcpy(s->history + PLC_HISTORY_LEN - s->buf_ptr, tmp, sizeof(pj_int16_t)*s->buf_ptr);
+ s->buf_ptr = 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+PJ_INLINE(int) amdf_pitch(int min_pitch, int max_pitch, pj_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, pj_int16_t amp[], int len)
+{
+ int i;
+ /*int overlap_len;*/
+ 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, pj_int16_t amp[], int len)
+{
+ /*pj_int16_t tmp[PLC_PITCH_OVERLAP_MAX];*/
+ int i;
+ int pitch_overlap;
+ float old_step;
+ float new_step;
+ float old_weight;
+ float new_weight;
+ float gain;
+ pj_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] = (pj_int16_t)(s->pitchbuf[s->pitch_offset]*gain);
+ 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 --------------------------------------------------------*/
+
+
+/*
+ * PJMEDIA specifics
+ */
+#include <pj/assert.h>
+#include <pj/pool.h>
+#include <pj/log.h>
+
+#define THIS_FILE "plc_steveu.c"
+
+struct steveu_plc
+{
+ plc_state_t state;
+ unsigned samples_per_frame;
+};
+
+void* pjmedia_plc_steveu_create(pj_pool_t *pool, unsigned c, unsigned f)
+{
+ struct steveu_plc *splc;
+
+ PJ_ASSERT_RETURN(c==8000, NULL);
+ PJ_UNUSED_ARG(c);
+
+ splc = pj_pool_alloc(pool, sizeof(struct steveu_plc));
+ plc_init(&splc->state);
+ splc->samples_per_frame = f;
+
+ return splc;
+}
+
+void pjmedia_plc_steveu_save(void *obj, pj_int16_t *samples)
+{
+ struct steveu_plc *splc = obj;
+ plc_rx(&splc->state, samples, splc->samples_per_frame);
+}
+
+void pjmedia_plc_steveu_generate(void *obj, pj_int16_t *samples)
+{
+ struct steveu_plc *splc = obj;
+ //PJ_LOG(5,(THIS_FILE, "PLC: generating lost frame"));
+ plc_fillin(&splc->state, samples, splc->samples_per_frame);
+}
+
+/*- End of file ------------------------------------------------------------*/
+