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Totally redo file formats

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2003-06-28 22:50:47 +00:00
parent 9befc69684
commit 3b78918878
13 changed files with 94 additions and 906 deletions
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108
formats/format_mp3.c
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@@ -34,14 +34,10 @@ struct ast_filestream {
void *reserved[AST_RESERVED_POINTERS];
/* This is what a filestream means to us */
int fd; /* Descriptor */
struct ast_channel *owner;
struct ast_filestream *next;
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;
};
@@ -68,14 +64,9 @@ static struct ast_filestream *mp3_open(int fd)
free(tmp);
return NULL;
}
tmp->next = glist;
glist = tmp;
tmp->fd = fd;
tmp->owner = NULL;
tmp->lasttimeout = -1;
tmp->last.tv_usec = 0;
tmp->last.tv_sec = 0;
tmp->adj = 0;
glistcnt++;
ast_pthread_mutex_unlock(&mp3_lock);
ast_update_use_count();
@@ -95,10 +86,7 @@ static struct ast_filestream *mp3_rewrite(int fd, char *comment)
free(tmp);
return NULL;
}
tmp->next = glist;
glist = tmp;
tmp->fd = fd;
tmp->owner = NULL;
glistcnt++;
ast_pthread_mutex_unlock(&mp3_lock);
ast_update_use_count();
@@ -107,71 +95,40 @@ static struct ast_filestream *mp3_rewrite(int fd, char *comment)
return tmp;
}
static struct ast_frame *mp3_read(struct ast_filestream *s)
{
return NULL;
}
static void mp3_close(struct ast_filestream *s)
{
struct ast_filestream *tmp, *tmpl = NULL;
if (ast_pthread_mutex_lock(&mp3_lock)) {
ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
return;
}
tmp = glist;
while(tmp) {
if (tmp == s) {
if (tmpl)
tmpl->next = tmp->next;
else
glist = tmp->next;
break;
}
tmpl = tmp;
tmp = tmp->next;
}
glistcnt--;
if (s->owner) {
s->owner->stream = NULL;
if (s->owner->streamid > -1)
ast_sched_del(s->owner->sched, s->owner->streamid);
s->owner->streamid = -1;
}
ast_pthread_mutex_unlock(&mp3_lock);
ast_update_use_count();
if (!tmp)
ast_log(LOG_WARNING, "Freeing a filestream we don't seem to own\n");
close(s->fd);
free(s);
}
static int ast_read_callback(void *data)
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;
struct ast_filestream *s = data;
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;
return NULL;
}
if (mp3_badheader(s->buf)) {
ast_log(LOG_WARNING, "Bad mp3 header\n");
return 0;
return NULL;
}
if ((size = mp3_framelen(s->buf)) < 0) {
ast_log(LOG_WARNING, "Unable to calculate frame size\n");
return 0;
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));
s->owner->streamid = -1;
return 0;
return NULL;
}
/* Send a frame from the file to the appropriate channel */
/* Read the data into the buffer */
@@ -183,48 +140,15 @@ static int ast_read_callback(void *data)
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) > 0 )
s->adj -= (ms - delay);
s->adj -= 2;
}
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 += s->adj;
delay *= 8;
if (delay < 1)
delay = 1;
/* Lastly, process the frame */
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;
}
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;
return 0;
}
static int mp3_play(struct ast_filestream *s)
{
ast_read_callback(s);
return 0;
*whennext = delay;
return &s->fr;
}
static int mp3_write(struct ast_filestream *fs, struct ast_frame *f)
@@ -270,8 +194,6 @@ int load_module()
return ast_format_register(name, exts, AST_FORMAT_MP3,
mp3_open,
mp3_rewrite,
mp3_apply,
mp3_play,
mp3_write,
mp3_seek,
mp3_trunc,
@@ -285,20 +207,6 @@ int load_module()
int unload_module()
{
struct ast_filestream *tmp, *tmpl;
if (ast_pthread_mutex_lock(&mp3_lock)) {
ast_log(LOG_WARNING, "Unable to lock mp3 list\n");
return -1;
}
tmp = glist;
while(tmp) {
if (tmp->owner)
ast_softhangup(tmp->owner, AST_SOFTHANGUP_APPUNLOAD);
tmpl = tmp;
tmp = tmp->next;
free(tmpl);
}
ast_pthread_mutex_unlock(&mp3_lock);
return ast_format_unregister(name);
}