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356 lines
8.6 KiB
356 lines
8.6 KiB
/*
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* Asterisk -- A telephony toolkit for Linux.
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*
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* Microsoft WAV File Format using libaudiofile
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*
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* Copyright (C) 1999, Mark Spencer
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*
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* Mark Spencer <markster@linux-support.net>
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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
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*/
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#include <asterisk/channel.h>
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#include <asterisk/file.h>
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#include <asterisk/logger.h>
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#include <asterisk/sched.h>
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#include <asterisk/module.h>
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#include <arpa/inet.h>
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#include <stdlib.h>
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#include <stdio.h>
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#include <unistd.h>
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#include <errno.h>
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#include <string.h>
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#include <pthread.h>
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#include <audiofile.h>
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/* Read 320 samples at a time, max */
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#define WAV_MAX_SIZE 320
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/* Fudge in milliseconds */
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#define WAV_FUDGE 2
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struct ast_filestream {
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/* First entry MUST be reserved for the channel type */
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void *reserved[AST_RESERVED_POINTERS];
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/* This is what a filestream means to us */
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int fd; /* Descriptor */
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/* Audio File */
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AFfilesetup afs;
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AFfilehandle af;
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int lasttimeout;
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struct ast_channel *owner;
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struct ast_filestream *next;
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struct ast_frame fr; /* Frame information */
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char waste[AST_FRIENDLY_OFFSET]; /* Buffer for sending frames, etc */
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short samples[WAV_MAX_SIZE];
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};
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static struct ast_filestream *glist = NULL;
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static pthread_mutex_t wav_lock = PTHREAD_MUTEX_INITIALIZER;
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static int glistcnt = 0;
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static char *name = "wav";
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static char *desc = "Microsoft WAV format (PCM/16, 8000Hz mono)";
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static char *exts = "wav";
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static struct ast_filestream *wav_open(int fd)
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{
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/* We don't have any header to read or anything really, but
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if we did, it would go here. We also might want to check
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and be sure it's a valid file. */
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struct ast_filestream *tmp;
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int notok = 0;
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int fmt, width;
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double rate;
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if ((tmp = malloc(sizeof(struct ast_filestream)))) {
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tmp->afs = afNewFileSetup();
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if (!tmp->afs) {
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ast_log(LOG_WARNING, "Unable to create file setup\n");
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free(tmp);
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return NULL;
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}
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afInitFileFormat(tmp->afs, AF_FILE_WAVE);
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tmp->af = afOpenFD(fd, "r", tmp->afs);
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if (!tmp->af) {
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afFreeFileSetup(tmp->afs);
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ast_log(LOG_WARNING, "Unable to open file descriptor\n");
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free(tmp);
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return NULL;
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}
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#if 0
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afGetFileFormat(tmp->af, &version);
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if (version != AF_FILE_WAVE) {
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ast_log(LOG_WARNING, "This is not a wave file (%d)\n", version);
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notok++;
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}
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#endif
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/* Read the format and make sure it's exactly what we seek. */
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if (afGetChannels(tmp->af, AF_DEFAULT_TRACK) != 1) {
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ast_log(LOG_WARNING, "Invalid number of channels %d. Should be mono (1)\n", afGetChannels(tmp->af, AF_DEFAULT_TRACK));
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notok++;
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}
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afGetSampleFormat(tmp->af, AF_DEFAULT_TRACK, &fmt, &width);
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if (fmt != AF_SAMPFMT_TWOSCOMP) {
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ast_log(LOG_WARNING, "Input file is not signed\n");
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notok++;
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}
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rate = afGetRate(tmp->af, AF_DEFAULT_TRACK);
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if ((rate < 7900) || (rate > 8100)) {
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ast_log(LOG_WARNING, "Rate %f is not close enough to 8000 Hz\n", rate);
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notok++;
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}
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if (width != 16) {
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ast_log(LOG_WARNING, "Input file is not 16-bit\n");
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notok++;
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}
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if (notok) {
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afCloseFile(tmp->af);
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afFreeFileSetup(tmp->afs);
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free(tmp);
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return NULL;
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}
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if (pthread_mutex_lock(&wav_lock)) {
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afCloseFile(tmp->af);
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afFreeFileSetup(tmp->afs);
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ast_log(LOG_WARNING, "Unable to lock wav list\n");
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free(tmp);
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return NULL;
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}
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tmp->next = glist;
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glist = tmp;
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tmp->fd = fd;
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tmp->owner = NULL;
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tmp->fr.data = tmp->samples;
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tmp->fr.frametype = AST_FRAME_VOICE;
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tmp->fr.subclass = AST_FORMAT_SLINEAR;
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/* datalen will vary for each frame */
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tmp->fr.src = name;
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tmp->fr.mallocd = 0;
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tmp->lasttimeout = -1;
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glistcnt++;
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pthread_mutex_unlock(&wav_lock);
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ast_update_use_count();
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}
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return tmp;
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}
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static struct ast_filestream *wav_rewrite(int fd, char *comment)
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{
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/* We don't have any header to read or anything really, but
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if we did, it would go here. We also might want to check
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and be sure it's a valid file. */
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struct ast_filestream *tmp;
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if ((tmp = malloc(sizeof(struct ast_filestream)))) {
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tmp->afs = afNewFileSetup();
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if (!tmp->afs) {
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ast_log(LOG_WARNING, "Unable to create file setup\n");
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free(tmp);
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return NULL;
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}
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/* WAV format */
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afInitFileFormat(tmp->afs, AF_FILE_WAVE);
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/* Mono */
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afInitChannels(tmp->afs, AF_DEFAULT_TRACK, 1);
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/* Signed linear, 16-bit */
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afInitSampleFormat(tmp->afs, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
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/* 8000 Hz */
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afInitRate(tmp->afs, AF_DEFAULT_TRACK, (double)8000.0);
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tmp->af = afOpenFD(fd, "w", tmp->afs);
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if (!tmp->af) {
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afFreeFileSetup(tmp->afs);
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ast_log(LOG_WARNING, "Unable to open file descriptor\n");
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free(tmp);
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return NULL;
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}
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if (pthread_mutex_lock(&wav_lock)) {
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ast_log(LOG_WARNING, "Unable to lock wav list\n");
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free(tmp);
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return NULL;
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}
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tmp->next = glist;
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glist = tmp;
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tmp->fd = fd;
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tmp->owner = NULL;
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tmp->lasttimeout = -1;
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glistcnt++;
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pthread_mutex_unlock(&wav_lock);
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ast_update_use_count();
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} else
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ast_log(LOG_WARNING, "Out of memory\n");
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return tmp;
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}
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static struct ast_frame *wav_read(struct ast_filestream *s)
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{
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return NULL;
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}
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static void wav_close(struct ast_filestream *s)
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{
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struct ast_filestream *tmp, *tmpl = NULL;
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if (pthread_mutex_lock(&wav_lock)) {
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ast_log(LOG_WARNING, "Unable to lock wav list\n");
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return;
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}
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tmp = glist;
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while(tmp) {
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if (tmp == s) {
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if (tmpl)
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tmpl->next = tmp->next;
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else
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glist = tmp->next;
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break;
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}
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tmpl = tmp;
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tmp = tmp->next;
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}
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glistcnt--;
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if (s->owner) {
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s->owner->stream = NULL;
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if (s->owner->streamid > -1)
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ast_sched_del(s->owner->sched, s->owner->streamid);
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s->owner->streamid = -1;
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}
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pthread_mutex_unlock(&wav_lock);
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ast_update_use_count();
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if (!tmp)
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ast_log(LOG_WARNING, "Freeing a filestream we don't seem to own\n");
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afCloseFile(tmp->af);
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afFreeFileSetup(tmp->afs);
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close(s->fd);
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free(s);
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}
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static int ast_read_callback(void *data)
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{
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u_int32_t delay = -1;
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int retval = 0;
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int res;
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struct ast_filestream *s = data;
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/* Send a frame from the file to the appropriate channel */
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if ((res = afReadFrames(s->af, AF_DEFAULT_TRACK, s->samples, sizeof(s->samples)/2)) < 1) {
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if (res)
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ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
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s->owner->streamid = -1;
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return 0;
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}
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/* Per 8 samples, one milisecond */
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delay = res / 8;
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s->fr.frametype = AST_FRAME_VOICE;
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s->fr.subclass = AST_FORMAT_SLINEAR;
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s->fr.offset = AST_FRIENDLY_OFFSET;
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s->fr.datalen = res * 2;
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s->fr.data = s->samples;
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s->fr.mallocd = 0;
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s->fr.timelen = delay;
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/* Unless there is no delay, we're going to exit out as soon as we
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have processed the current frame. */
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/* If there is a delay, lets schedule the next event */
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if (delay != s->lasttimeout) {
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/* We'll install the next timeout now. */
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s->owner->streamid = ast_sched_add(s->owner->sched,
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delay,
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ast_read_callback, s);
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s->lasttimeout = delay;
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} else {
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/* Just come back again at the same time */
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retval = -1;
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}
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/* Lastly, process the frame */
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if (ast_write(s->owner, &s->fr)) {
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ast_log(LOG_WARNING, "Failed to write frame\n");
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s->owner->streamid = -1;
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return 0;
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}
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return retval;
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}
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static int wav_apply(struct ast_channel *c, struct ast_filestream *s)
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{
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/* Select our owner for this stream, and get the ball rolling. */
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s->owner = c;
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ast_read_callback(s);
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return 0;
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}
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static int wav_write(struct ast_filestream *fs, struct ast_frame *f)
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{
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int res;
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if (f->frametype != AST_FRAME_VOICE) {
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ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
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return -1;
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}
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if (f->subclass != AST_FORMAT_SLINEAR) {
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ast_log(LOG_WARNING, "Asked to write non-signed linear frame (%d)!\n", f->subclass);
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return -1;
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}
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if ((res = afWriteFrames(fs->af, AF_DEFAULT_TRACK, f->data, f->datalen/2)) != f->datalen/2) {
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ast_log(LOG_WARNING, "Unable to write frame: res=%d (%s)\n", res, strerror(errno));
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return -1;
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}
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return 0;
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}
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static char *wav_getcomment(struct ast_filestream *s)
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{
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return NULL;
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}
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int load_module()
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{
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return ast_format_register(name, exts, AST_FORMAT_SLINEAR,
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wav_open,
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wav_rewrite,
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wav_apply,
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wav_write,
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wav_read,
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wav_close,
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wav_getcomment);
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}
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int unload_module()
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{
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struct ast_filestream *tmp, *tmpl;
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if (pthread_mutex_lock(&wav_lock)) {
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ast_log(LOG_WARNING, "Unable to lock wav list\n");
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return -1;
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}
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tmp = glist;
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while(tmp) {
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if (tmp->owner)
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ast_softhangup(tmp->owner);
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tmpl = tmp;
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tmp = tmp->next;
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free(tmpl);
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}
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pthread_mutex_unlock(&wav_lock);
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return ast_format_unregister(name);
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}
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int usecount()
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{
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int res;
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if (pthread_mutex_lock(&wav_lock)) {
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ast_log(LOG_WARNING, "Unable to lock wav list\n");
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return -1;
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}
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res = glistcnt;
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pthread_mutex_unlock(&wav_lock);
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return res;
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}
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char *description()
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{
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return desc;
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}
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