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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1486 65c4cc65-6c06-0410-ace0-fbb531ad65f3
237 lines
5.3 KiB
C
Executable File
237 lines
5.3 KiB
C
Executable File
/*
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* Asterisk -- A telephony toolkit for Linux.
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*
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* Everybody's favorite format: MP3 Files! Yay!
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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/lock.h>
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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 <netinet/in.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 <sys/time.h>
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#include "../channels/adtranvofr.h"
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#define MAX_FRAME_SIZE 1441
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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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struct ast_frame fr; /* Frame representation of buf */
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char offset[AST_FRIENDLY_OFFSET];
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unsigned char buf[MAX_FRAME_SIZE * 2];
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int pos;
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struct timeval last;
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};
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#if 0
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static struct ast_filestream *glist = NULL;
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#endif
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static ast_mutex_t mp3_lock = AST_MUTEX_INITIALIZER;
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static int glistcnt = 0;
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static char *name = "mp3";
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static char *desc = "MPEG-1,2 Layer 3 File Format Support";
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static char *exts = "mp3|mpeg3";
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#include "../codecs/mp3anal.h"
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static struct ast_filestream *mp3_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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if ((tmp = malloc(sizeof(struct ast_filestream)))) {
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if (ast_mutex_lock(&mp3_lock)) {
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ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
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free(tmp);
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return NULL;
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}
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tmp->fd = fd;
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tmp->last.tv_usec = 0;
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tmp->last.tv_sec = 0;
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glistcnt++;
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ast_mutex_unlock(&mp3_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 *mp3_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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if (ast_mutex_lock(&mp3_lock)) {
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ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
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free(tmp);
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return NULL;
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}
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tmp->fd = fd;
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glistcnt++;
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ast_mutex_unlock(&mp3_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 void mp3_close(struct ast_filestream *s)
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{
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if (ast_mutex_lock(&mp3_lock)) {
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ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
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return;
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}
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glistcnt--;
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ast_mutex_unlock(&mp3_lock);
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ast_update_use_count();
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close(s->fd);
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free(s);
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}
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static struct ast_frame *mp3_read(struct ast_filestream *s, int *whennext)
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{
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/* XXX Don't assume frames are this size XXX */
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u_int32_t delay = -1;
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int res;
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int size;
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if ((res = read(s->fd, s->buf , 4)) != 4) {
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ast_log(LOG_WARNING, "Short read (%d of 4 bytes) (%s)!\n", res, strerror(errno));
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return NULL;
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}
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if (mp3_badheader(s->buf)) {
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ast_log(LOG_WARNING, "Bad mp3 header\n");
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return NULL;
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}
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if ((size = mp3_framelen(s->buf)) < 0) {
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ast_log(LOG_WARNING, "Unable to calculate frame size\n");
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return NULL;
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}
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if ((res = read(s->fd, s->buf + 4 , size - 4)) != size - 4) {
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ast_log(LOG_WARNING, "Short read (%d of %d bytes) (%s)!\n", res, size - 4, strerror(errno));
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return NULL;
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}
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/* Send a frame from the file to the appropriate channel */
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/* Read the data into the buffer */
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s->fr.offset = AST_FRIENDLY_OFFSET;
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s->fr.frametype = AST_FRAME_VOICE;
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s->fr.subclass = AST_FORMAT_MP3;
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s->fr.mallocd = 0;
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s->fr.src = name;
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s->fr.datalen = size;
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s->fr.data = s->buf;
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delay = mp3_samples(s->buf) * 1000 / mp3_samplerate(s->buf);
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s->fr.samples = delay * 8;
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#if 0
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ast_log(LOG_DEBUG, "delay is %d, adjusting by %d, as last was %d\n", delay, s->adj, ms);
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#endif
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delay *= 8;
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if (delay < 1)
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delay = 1;
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*whennext = delay;
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return &s->fr;
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}
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static int mp3_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_MP3) {
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ast_log(LOG_WARNING, "Asked to write non-mp3 frame!\n");
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return -1;
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}
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if ((res = write(fs->fd, f->data, f->datalen)) != f->datalen) {
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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 int mp3_seek(struct ast_filestream *fs, long sample_offset, int whence)
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{
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return -1;
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}
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static int mp3_trunc(struct ast_filestream *fs)
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{
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return -1;
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}
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static long mp3_tell(struct ast_filestream *fs)
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{
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return -1;
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}
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static char *mp3_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_MP3,
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mp3_open,
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mp3_rewrite,
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mp3_write,
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mp3_seek,
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mp3_trunc,
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mp3_tell,
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mp3_read,
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mp3_close,
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mp3_getcomment);
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}
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int unload_module()
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{
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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 (ast_mutex_lock(&mp3_lock)) {
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ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
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return -1;
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}
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res = glistcnt;
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ast_mutex_unlock(&mp3_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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char *key()
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{
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return ASTERISK_GPL_KEY;
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}
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