Version 0.1.1 from FTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
1999-12-16 13:44:30 +00:00
parent d73d60a578
commit 0ed9477395
13 changed files with 1750 additions and 68 deletions

View File

@@ -3,7 +3,7 @@
*
* Everybody's favorite format: MP3 Files! Yay!
*
* Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
* Copyright (C) 1999, Mark Spencer
*
* Mark Spencer <markster@linux-support.net>
*
@@ -26,8 +26,7 @@
#include <sys/time.h>
#include "../channels/adtranvofr.h"
#define MP3_MAX_SIZE 1400
#define MAX_FRAME_SIZE 1441
struct ast_filestream {
/* First entry MUST be reserved for the channel type */
@@ -36,9 +35,13 @@ struct ast_filestream {
int fd; /* Descriptor */
struct ast_channel *owner;
struct ast_filestream *next;
struct ast_frame *fr; /* Frame representation of buf */
char buf[sizeof(struct ast_frame) + MP3_MAX_SIZE + AST_FRIENDLY_OFFSET]; /* Buffer for sending frames, etc */
struct ast_frame fr; /* Frame representation of buf */
char offset[AST_FRIENDLY_OFFSET];
unsigned char buf[MAX_FRAME_SIZE * 2];
int lasttimeout;
int pos;
int adj;
struct timeval last;
};
@@ -47,16 +50,10 @@ static pthread_mutex_t mp3_lock = PTHREAD_MUTEX_INITIALIZER;
static int glistcnt = 0;
static char *name = "mp3";
static char *desc = "MPEG-2 Layer 3 File Format Support";
static char *desc = "MPEG-1,2 Layer 3 File Format Support";
static char *exts = "mp3|mpeg3";
#if 0
#define MP3_FRAMELEN 417
#else
#define MP3_FRAMELEN 400
#endif
#define MP3_OUTPUTLEN 2304 /* Bytes */
#define MP3_TIMELEN ((MP3_OUTPUTLEN * 1000 / 16000) )
#include "../codecs/mp3anal.h"
static struct ast_filestream *mp3_open(int fd)
{
@@ -74,13 +71,10 @@ static struct ast_filestream *mp3_open(int fd)
glist = tmp;
tmp->fd = fd;
tmp->owner = NULL;
tmp->fr = (struct ast_frame *)tmp->buf;
tmp->fr->data = tmp->buf + sizeof(struct ast_frame);
tmp->fr->frametype = AST_FRAME_VOICE;
tmp->fr->subclass = AST_FORMAT_MP3;
/* datalen will vary for each frame */
tmp->fr->src = name;
tmp->fr->mallocd = 0;
tmp->lasttimeout = -1;
tmp->last.tv_usec = 0;
tmp->last.tv_sec = 0;
tmp->adj = 0;
glistcnt++;
pthread_mutex_unlock(&mp3_lock);
ast_update_use_count();
@@ -104,7 +98,6 @@ static struct ast_filestream *mp3_rewrite(int fd, char *comment)
glist = tmp;
tmp->fd = fd;
tmp->owner = NULL;
tmp->fr = NULL;
glistcnt++;
pthread_mutex_unlock(&mp3_lock);
ast_update_use_count();
@@ -155,28 +148,67 @@ static void mp3_close(struct ast_filestream *s)
static int ast_read_callback(void *data)
{
/* XXX Don't assume frames are this size XXX */
u_int16_t size=MP3_FRAMELEN;
u_int32_t delay = -1;
int res;
struct ast_filestream *s = data;
/* Send a frame from the file to the appropriate channel */
/* Read the data into the buffer */
s->fr->offset = AST_FRIENDLY_OFFSET;
s->fr->datalen = size;
s->fr->data = s->buf + sizeof(struct ast_frame) + AST_FRIENDLY_OFFSET;
if ((res = read(s->fd, s->fr->data , size)) != size) {
ast_log(LOG_WARNING, "Short read (%d of %d bytes) (%s)!\n", res, size, strerror(errno));
int size;
int ms=0;
struct timeval tv;
if ((res = read(s->fd, s->buf , 4)) != 4) {
ast_log(LOG_WARNING, "Short read (%d of 4 bytes) (%s)!\n", res, strerror(errno));
s->owner->streamid = -1;
return 0;
}
delay = MP3_TIMELEN;
s->fr->timelen = delay;
if (mp3_badheader(s->buf)) {
ast_log(LOG_WARNING, "Bad mp3 header\n");
return 0;
}
if ((size = mp3_framelen(s->buf)) < 0) {
ast_log(LOG_WARNING, "Unable to calculate frame size\n");
return 0;
}
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));
s->owner->streamid = -1;
return 0;
}
/* 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);
if (s->last.tv_sec || s->last.tv_usec) {
/* To keep things running smoothly, we watch how close we're coming */
gettimeofday(&tv, NULL);
ms = ((tv.tv_usec - s->last.tv_usec) / 1000 + (tv.tv_sec - s->last.tv_sec) * 1000);
/* If we're within 2 milliseconds, that's close enough */
if ((ms - delay) * (ms - delay) > 4)
s->adj -= (ms - delay);
}
s->fr.timelen = delay;
#if 0
ast_log(LOG_DEBUG, "delay is %d, adjusting by %d, as last was %d\n", delay, s->adj, ms);
#endif
delay += s->adj;
if (delay < 1)
delay = 1;
/* Lastly, process the frame */
if (ast_write(s->owner, s->fr)) {
if (ast_write(s->owner, &s->fr)) {
ast_log(LOG_WARNING, "Failed to write frame\n");
s->owner->streamid = -1;
return 0;
}
gettimeofday(&s->last, NULL);
if (s->lasttimeout != delay) {
s->owner->streamid = ast_sched_add(s->owner->sched, delay, ast_read_callback, s);
s->lasttimeout = delay;
return 0;
}
return -1;
}
@@ -184,7 +216,6 @@ static int mp3_apply(struct ast_channel *c, struct ast_filestream *s)
{
/* Select our owner for this stream, and get the ball rolling. */
s->owner = c;
s->owner->streamid = ast_sched_add(s->owner->sched, MP3_TIMELEN, ast_read_callback, s);
ast_read_callback(s);
return 0;
}
@@ -192,10 +223,6 @@ static int mp3_apply(struct ast_channel *c, struct ast_filestream *s)
static int mp3_write(struct ast_filestream *fs, struct ast_frame *f)
{
int res;
if (fs->fr) {
ast_log(LOG_WARNING, "Asked to write on a read stream??\n");
return -1;
}
if (f->frametype != AST_FRAME_VOICE) {
ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
return -1;
@@ -211,7 +238,7 @@ static int mp3_write(struct ast_filestream *fs, struct ast_frame *f)
return 0;
}
char *mp3_getcomment(struct ast_filestream *s)
static char *mp3_getcomment(struct ast_filestream *s)
{
return NULL;
}