Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
251 lines
7.1 KiB
C
251 lines
7.1 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Written by Steve Underwood <steveu@coppice.org>
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*
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* Copyright (C) 2004 Steve Underwood
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*
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* All rights reserved.
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*
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* This version may be optionally licenced under the GNU LGPL licence.
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*
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* A license has been granted to Digium (via disclaimer) for the use of
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* this code.
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*/
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/*! \file
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*
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* \brief SpanDSP - a series of DSP components for telephony
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*
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* \author Steve Underwood <steveu@coppice.org>
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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ASTERISK_REGISTER_FILE()
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#include <math.h>
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#include "asterisk/plc.h"
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#if !defined(FALSE)
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#define FALSE 0
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#endif
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#if !defined(TRUE)
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#define TRUE (!FALSE)
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#endif
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#if !defined(INT16_MAX)
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#define INT16_MAX (32767)
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#define INT16_MIN (-32767-1)
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#endif
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/* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
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#define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */
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#define ms_to_samples(t) (((t)*DEFAULT_SAMPLE_RATE)/1000)
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static inline int16_t fsaturate(double damp)
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{
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if (damp > 32767.0)
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return INT16_MAX;
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if (damp < -32768.0)
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return INT16_MIN;
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return (int16_t) rint(damp);
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}
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static void save_history(plc_state_t *s, int16_t *buf, int len)
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{
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if (len >= PLC_HISTORY_LEN) {
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/* Just keep the last part of the new data, starting at the beginning of the buffer */
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memcpy(s->history, buf + len - PLC_HISTORY_LEN, sizeof(int16_t) * PLC_HISTORY_LEN);
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s->buf_ptr = 0;
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return;
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}
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if (s->buf_ptr + len > PLC_HISTORY_LEN) {
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/* Wraps around - must break into two sections */
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memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr));
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len -= (PLC_HISTORY_LEN - s->buf_ptr);
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memcpy(s->history, buf + (PLC_HISTORY_LEN - s->buf_ptr), sizeof(int16_t)*len);
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s->buf_ptr = len;
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return;
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}
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/* Can use just one section */
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memcpy(s->history + s->buf_ptr, buf, sizeof(int16_t)*len);
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s->buf_ptr += len;
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}
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/*- End of function --------------------------------------------------------*/
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static void normalise_history(plc_state_t *s)
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{
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int16_t tmp[PLC_HISTORY_LEN];
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if (s->buf_ptr == 0)
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return;
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memcpy(tmp, s->history, sizeof(int16_t)*s->buf_ptr);
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memcpy(s->history, s->history + s->buf_ptr, sizeof(int16_t) * (PLC_HISTORY_LEN - s->buf_ptr));
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memcpy(s->history + PLC_HISTORY_LEN - s->buf_ptr, tmp, sizeof(int16_t) * s->buf_ptr);
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s->buf_ptr = 0;
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}
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/*- End of function --------------------------------------------------------*/
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static int __inline__ amdf_pitch(int min_pitch, int max_pitch, int16_t amp[], int len)
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{
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int i;
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int j;
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int acc;
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int min_acc;
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int pitch;
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pitch = min_pitch;
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min_acc = INT_MAX;
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for (i = max_pitch; i <= min_pitch; i++) {
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acc = 0;
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for (j = 0; j < len; j++)
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acc += abs(amp[i + j] - amp[j]);
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if (acc < min_acc) {
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min_acc = acc;
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pitch = i;
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}
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}
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return pitch;
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}
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/*- End of function --------------------------------------------------------*/
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int plc_rx(plc_state_t *s, int16_t amp[], int len)
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{
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int i;
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int pitch_overlap;
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float old_step;
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float new_step;
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float old_weight;
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float new_weight;
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float gain;
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if (s->missing_samples) {
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/* Although we have a real signal, we need to smooth it to fit well
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with the synthetic signal we used for the previous block */
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/* The start of the real data is overlapped with the next 1/4 cycle
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of the synthetic data. */
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pitch_overlap = s->pitch >> 2;
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if (pitch_overlap > len)
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pitch_overlap = len;
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gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
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if (gain < 0.0)
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gain = 0.0;
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new_step = 1.0/pitch_overlap;
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old_step = new_step*gain;
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new_weight = new_step;
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old_weight = (1.0 - new_step)*gain;
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for (i = 0; i < pitch_overlap; i++) {
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amp[i] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[i]);
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if (++s->pitch_offset >= s->pitch)
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s->pitch_offset = 0;
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new_weight += new_step;
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old_weight -= old_step;
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if (old_weight < 0.0)
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old_weight = 0.0;
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}
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s->missing_samples = 0;
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}
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save_history(s, amp, len);
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return len;
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}
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/*- End of function --------------------------------------------------------*/
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int plc_fillin(plc_state_t *s, int16_t amp[], int len)
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{
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int i;
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int pitch_overlap;
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float old_step;
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float new_step;
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float old_weight;
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float new_weight;
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float gain;
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int orig_len;
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orig_len = len;
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if (s->missing_samples == 0) {
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/* As the gap in real speech starts we need to assess the last known pitch,
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and prepare the synthetic data we will use for fill-in */
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normalise_history(s);
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s->pitch = amdf_pitch(PLC_PITCH_MIN, PLC_PITCH_MAX, s->history + PLC_HISTORY_LEN - CORRELATION_SPAN - PLC_PITCH_MIN, CORRELATION_SPAN);
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/* We overlap a 1/4 wavelength */
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pitch_overlap = s->pitch >> 2;
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/* Cook up a single cycle of pitch, using a single of the real signal with 1/4
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cycle OLA'ed to make the ends join up nicely */
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/* The first 3/4 of the cycle is a simple copy */
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for (i = 0; i < s->pitch - pitch_overlap; i++)
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s->pitchbuf[i] = s->history[PLC_HISTORY_LEN - s->pitch + i];
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/* The last 1/4 of the cycle is overlapped with the end of the previous cycle */
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new_step = 1.0/pitch_overlap;
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new_weight = new_step;
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for ( ; i < s->pitch; i++) {
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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;
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new_weight += new_step;
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}
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/* We should now be ready to fill in the gap with repeated, decaying cycles
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of what is in pitchbuf */
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/* We need to OLA the first 1/4 wavelength of the synthetic data, to smooth
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it into the previous real data. To avoid the need to introduce a delay
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in the stream, reverse the last 1/4 wavelength, and OLA with that. */
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gain = 1.0;
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new_step = 1.0 / pitch_overlap;
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old_step = new_step;
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new_weight = new_step;
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old_weight = 1.0 - new_step;
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for (i = 0; i < pitch_overlap; i++) {
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amp[i] = fsaturate(old_weight * s->history[PLC_HISTORY_LEN - 1 - i] + new_weight * s->pitchbuf[i]);
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new_weight += new_step;
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old_weight -= old_step;
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if (old_weight < 0.0)
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old_weight = 0.0;
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}
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s->pitch_offset = i;
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} else {
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gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT;
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i = 0;
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}
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for ( ; gain > 0.0 && i < len; i++) {
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amp[i] = s->pitchbuf[s->pitch_offset] * gain;
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gain -= ATTENUATION_INCREMENT;
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if (++s->pitch_offset >= s->pitch)
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s->pitch_offset = 0;
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}
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for ( ; i < len; i++)
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amp[i] = 0;
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s->missing_samples += orig_len;
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save_history(s, amp, len);
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return len;
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}
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/*- End of function --------------------------------------------------------*/
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plc_state_t *plc_init(plc_state_t *s)
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{
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memset(s, 0, sizeof(*s));
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return s;
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}
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/*- End of function --------------------------------------------------------*/
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/*- End of file ------------------------------------------------------------*/
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