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Remove really broke MP3 stuff in favor of G.726 in the near future

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2003-11-04 02:40:09 +00:00
parent 7ec16a6af7
commit 1e95c3a4ac
58 changed files with 27 additions and 15549 deletions
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2
formats/Makefile
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@@ -11,7 +11,7 @@
# the GNU General Public License
#
FORMAT_LIBS=format_gsm.so format_wav.so format_mp3.so \
FORMAT_LIBS=format_gsm.so format_wav.so \
format_wav_gsm.so format_vox.so format_pcm.so format_g729.so \
format_pcm_alaw.so format_h263.so
FORMAT_LIBS+=format_jpeg.so

236
formats/format_mp3.c
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@@ -1,236 +0,0 @@
/*
* Asterisk -- A telephony toolkit for Linux.
*
* Everybody's favorite format: MP3 Files! Yay!
*
* Copyright (C) 1999, Mark Spencer
*
* Mark Spencer <markster@linux-support.net>
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*/
#include <asterisk/lock.h>
#include <asterisk/channel.h>
#include <asterisk/file.h>
#include <asterisk/logger.h>
#include <asterisk/sched.h>
#include <asterisk/module.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <stdlib.h>
#include <stdio.h>
#include <unistd.h>
#include <errno.h>
#include <string.h>
#include <pthread.h>
#include <sys/time.h>
#include "../channels/adtranvofr.h"
#define MAX_FRAME_SIZE 1441
struct ast_filestream {
/* First entry MUST be reserved for the channel type */
void *reserved[AST_RESERVED_POINTERS];
/* This is what a filestream means to us */
int fd; /* Descriptor */
struct ast_frame fr; /* Frame representation of buf */
char offset[AST_FRIENDLY_OFFSET];
unsigned char buf[MAX_FRAME_SIZE * 2];
int pos;
struct timeval last;
};
#if 0
static struct ast_filestream *glist = NULL;
#endif
static ast_mutex_t mp3_lock = AST_MUTEX_INITIALIZER;
static int glistcnt = 0;
static char *name = "mp3";
static char *desc = "MPEG-1,2 Layer 3 File Format Support";
static char *exts = "mp3|mpeg3";
#include "../codecs/mp3anal.h"
static struct ast_filestream *mp3_open(int fd)
{
/* We don't have any header to read or anything really, but
if we did, it would go here. We also might want to check
and be sure it's a valid file. */
struct ast_filestream *tmp;
if ((tmp = malloc(sizeof(struct ast_filestream)))) {
if (ast_mutex_lock(&mp3_lock)) {
ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
free(tmp);
return NULL;
}
tmp->fd = fd;
tmp->last.tv_usec = 0;
tmp->last.tv_sec = 0;
glistcnt++;
ast_mutex_unlock(&mp3_lock);
ast_update_use_count();
}
return tmp;
}
static struct ast_filestream *mp3_rewrite(int fd, char *comment)
{
/* We don't have any header to read or anything really, but
if we did, it would go here. We also might want to check
and be sure it's a valid file. */
struct ast_filestream *tmp;
if ((tmp = malloc(sizeof(struct ast_filestream)))) {
if (ast_mutex_lock(&mp3_lock)) {
ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
free(tmp);
return NULL;
}
tmp->fd = fd;
glistcnt++;
ast_mutex_unlock(&mp3_lock);
ast_update_use_count();
} else
ast_log(LOG_WARNING, "Out of memory\n");
return tmp;
}
static void mp3_close(struct ast_filestream *s)
{
if (ast_mutex_lock(&mp3_lock)) {
ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
return;
}
glistcnt--;
ast_mutex_unlock(&mp3_lock);
ast_update_use_count();
close(s->fd);
free(s);
}
static struct ast_frame *mp3_read(struct ast_filestream *s, int *whennext)
{
/* XXX Don't assume frames are this size XXX */
u_int32_t delay = -1;
int res;
int size;
if ((res = read(s->fd, s->buf , 4)) != 4) {
ast_log(LOG_WARNING, "Short read (%d of 4 bytes) (%s)!\n", res, strerror(errno));
return NULL;
}
if (mp3_badheader(s->buf)) {
ast_log(LOG_WARNING, "Bad mp3 header\n");
return NULL;
}
if ((size = mp3_framelen(s->buf)) < 0) {
ast_log(LOG_WARNING, "Unable to calculate frame size\n");
return NULL;
}
if ((res = read(s->fd, s->buf + 4 , size - 4)) != size - 4) {
ast_log(LOG_WARNING, "Short read (%d of %d bytes) (%s)!\n", res, size - 4, strerror(errno));
return NULL;
}
/* Send a frame from the file to the appropriate channel */
/* Read the data into the buffer */
s->fr.offset = AST_FRIENDLY_OFFSET;
s->fr.frametype = AST_FRAME_VOICE;
s->fr.subclass = AST_FORMAT_MP3;
s->fr.mallocd = 0;
s->fr.src = name;
s->fr.datalen = size;
s->fr.data = s->buf;
delay = mp3_samples(s->buf) * 1000 / mp3_samplerate(s->buf);
s->fr.samples = delay * 8;
#if 0
ast_log(LOG_DEBUG, "delay is %d, adjusting by %d, as last was %d\n", delay, s->adj, ms);
#endif
delay *= 8;
if (delay < 1)
delay = 1;
*whennext = delay;
return &s->fr;
}
static int mp3_write(struct ast_filestream *fs, struct ast_frame *f)
{
int res;
if (f->frametype != AST_FRAME_VOICE) {
ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
return -1;
}
if (f->subclass != AST_FORMAT_MP3) {
ast_log(LOG_WARNING, "Asked to write non-mp3 frame!\n");
return -1;
}
if ((res = write(fs->fd, f->data, f->datalen)) != f->datalen) {
ast_log(LOG_WARNING, "Unable to write frame: res=%d (%s)\n", res, strerror(errno));
return -1;
}
return 0;
}
static int mp3_seek(struct ast_filestream *fs, long sample_offset, int whence)
{
return -1;
}
static int mp3_trunc(struct ast_filestream *fs)
{
return -1;
}
static long mp3_tell(struct ast_filestream *fs)
{
return -1;
}
static char *mp3_getcomment(struct ast_filestream *s)
{
return NULL;
}
int load_module()
{
return ast_format_register(name, exts, AST_FORMAT_MP3,
mp3_open,
mp3_rewrite,
mp3_write,
mp3_seek,
mp3_trunc,
mp3_tell,
mp3_read,
mp3_close,
mp3_getcomment);
}
int unload_module()
{
return ast_format_unregister(name);
}
int usecount()
{
int res;
if (ast_mutex_lock(&mp3_lock)) {
ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
return -1;
}
res = glistcnt;
ast_mutex_unlock(&mp3_lock);
return res;
}
char *description()
{
return desc;
}
char *key()
{
return ASTERISK_GPL_KEY;
}