Now really checked in the new PLC software!
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@724 74dad513-b988-da41-8d7b-12977e46ad98
diff --git a/pjmedia/src/pjmedia/plc_steveu.c b/pjmedia/src/pjmedia/plc_steveu.c
new file mode 100644
index 0000000..3326b74
--- /dev/null
+++ b/pjmedia/src/pjmedia/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 ------------------------------------------------------------*/
+
diff --git a/pjmedia/src/pjmedia/plc_steveu.h b/pjmedia/src/pjmedia/plc_steveu.h
new file mode 100644
index 0000000..483e774
--- /dev/null
+++ b/pjmedia/src/pjmedia/plc_steveu.h
@@ -0,0 +1,153 @@
+/*! \file
+ * \brief SpanDSP - a series of DSP components for telephony
+ *
+ * plc.h
+ *
+ * \author 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.
+ *
+ * A license has been granted to Digium (via disclaimer) for the use of
+ * this code.
+ */
+
+
+#if !defined(_PLC_H_)
+#define _PLC_H_
+
+
+/*! \page plc_page Packet loss concealment
+\section plc_page_sec_1 What does it do?
+The packet loss concealment module provides a suitable synthetic fill-in signal,
+to minimise the audible effect of lost packets in VoIP applications. It is not
+tied to any particular codec, and could be used with almost any codec which does not
+specify its own procedure for packet loss concealment.
+
+Where a codec specific concealment procedure exists, the algorithm is usually built
+around knowledge of the characteristics of the particular codec. It will, therefore,
+generally give better results for that particular codec than this generic concealer will.
+
+\section plc_page_sec_2 How does it work?
+While good packets are being received, the plc_rx() routine keeps a record of the trailing
+section of the known speech signal. If a packet is missed, plc_fillin() is called to produce
+a synthetic replacement for the real speech signal. The average mean difference function
+(AMDF) is applied to the last known good signal, to determine its effective pitch.
+Based on this, the last pitch period of signal is saved. Essentially, this cycle of speech
+will be repeated over and over until the real speech resumes. However, several refinements
+are needed to obtain smooth pleasant sounding results.
+
+- The two ends of the stored cycle of speech will not always fit together smoothly. This can
+ cause roughness, or even clicks, at the joins between cycles. To soften this, the
+ 1/4 pitch period of real speech preceeding the cycle to be repeated is blended with the last
+ 1/4 pitch period of the cycle to be repeated, using an overlap-add (OLA) technique (i.e.
+ in total, the last 5/4 pitch periods of real speech are used).
+
+- The start of the synthetic speech will not always fit together smoothly with the tail of
+ real speech passed on before the erasure was identified. Ideally, we would like to modify
+ the last 1/4 pitch period of the real speech, to blend it into the synthetic speech. However,
+ it is too late for that. We could have delayed the real speech a little, but that would
+ require more buffer manipulation, and hurt the efficiency of the no-lost-packets case
+ (which we hope is the dominant case). Instead we use a degenerate form of OLA to modify
+ the start of the synthetic data. The last 1/4 pitch period of real speech is time reversed,
+ and OLA is used to blend it with the first 1/4 pitch period of synthetic speech. The result
+ seems quite acceptable.
+
+- As we progress into the erasure, the chances of the synthetic signal being anything like
+ correct steadily fall. Therefore, the volume of the synthesized signal is made to decay
+ linearly, such that after 50ms of missing audio it is reduced to silence.
+
+- When real speech resumes, an extra 1/4 pitch period of sythetic speech is blended with the
+ start of the real speech. If the erasure is small, this smoothes the transition. If the erasure
+ is long, and the synthetic signal has faded to zero, the blending softens the start up of the
+ real signal, avoiding a kind of "click" or "pop" effect that might occur with a sudden onset.
+
+\section plc_page_sec_3 How do I use it?
+Before audio is processed, call plc_init() to create an instance of the packet loss
+concealer. For each received audio packet that is acceptable (i.e. not including those being
+dropped for being too late) call plc_rx() to record the content of the packet. Note this may
+modify the packet a little after a period of packet loss, to blend real synthetic data smoothly.
+When a real packet is not available in time, call plc_fillin() to create a sythetic substitute.
+That's it!
+*/
+
+/*! Minimum allowed pitch (66 Hz) */
+#define PLC_PITCH_MIN 120
+/*! Maximum allowed pitch (200 Hz) */
+#define PLC_PITCH_MAX 40
+/*! Maximum pitch OLA window */
+#define PLC_PITCH_OVERLAP_MAX (PLC_PITCH_MIN >> 2)
+/*! The length over which the AMDF function looks for similarity (20 ms) */
+#define CORRELATION_SPAN 160
+/*! 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. */
+#define PLC_HISTORY_LEN (CORRELATION_SPAN + PLC_PITCH_MIN)
+
+typedef struct
+{
+ /*! Consecutive erased samples */
+ int missing_samples;
+ /*! Current offset into pitch period */
+ int pitch_offset;
+ /*! Pitch estimate */
+ int pitch;
+ /*! Buffer for a cycle of speech */
+ float pitchbuf[PLC_PITCH_MIN];
+ /*! History buffer */
+ pj_int16_t history[PLC_HISTORY_LEN];
+ /*! Current pointer into the history buffer */
+ int buf_ptr;
+} plc_state_t;
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*! Process a block of received audio samples.
+ \brief Process a block of received audio samples.
+ \param s The packet loss concealer context.
+ \param amp The audio sample buffer.
+ \param len The number of samples in the buffer.
+ \return The number of samples in the buffer. */
+int plc_rx(plc_state_t *s, pj_int16_t amp[], int len);
+
+/*! Fill-in a block of missing audio samples.
+ \brief Fill-in a block of missing audio samples.
+ \param s The packet loss concealer context.
+ \param amp The audio sample buffer.
+ \param len The number of samples to be synthesised.
+ \return The number of samples synthesized. */
+int plc_fillin(plc_state_t *s, pj_int16_t amp[], int len);
+
+/*! Process a block of received V.29 modem audio samples.
+ \brief Process a block of received V.29 modem audio samples.
+ \param s The packet loss concealer context.
+ \return A pointer to the he packet loss concealer context. */
+plc_state_t *plc_init(plc_state_t *s);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/
+