From 8b0c007ad990aa27d9868da49215fd1076ac77cc Mon Sep 17 00:00:00 2001 From: kpfleming Date: Mon, 21 Aug 2006 02:11:39 +0000 Subject: merge new_loader_completion branch, including (at least): - restructured build tree and makefiles to eliminate recursion problems - support for embedded modules - support for static builds - simpler cross-compilation support - simpler module/loader interface (no exported symbols) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b --- main/plc.c | 251 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 251 insertions(+) create mode 100644 main/plc.c (limited to 'main/plc.c') diff --git a/main/plc.c b/main/plc.c new file mode 100644 index 000000000..336a99030 --- /dev/null +++ b/main/plc.c @@ -0,0 +1,251 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Written by Steve Underwood + * + * 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 + */ + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#include +#include +#include +#include + +#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 ------------------------------------------------------------*/ -- cgit v1.2.3