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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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96
formats/format_wav.c
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@@ -41,7 +41,6 @@ struct ast_filestream {
int fd; /* Descriptor */
int bytes;
int needsgain;
struct ast_channel *owner;
struct ast_frame fr; /* Frame information */
char waste[AST_FRIENDLY_OFFSET]; /* Buffer for sending frames, etc */
char empty; /* Empty character */
@@ -50,12 +49,9 @@ struct ast_filestream {
int lasttimeout;
int maxlen;
struct timeval last;
int adj;
struct ast_filestream *next;
};
static struct ast_filestream *glist = NULL;
static pthread_mutex_t wav_lock = AST_MUTEX_INITIALIZER;
static int glistcnt = 0;
@@ -321,10 +317,7 @@ static struct ast_filestream *wav_open(int fd)
free(tmp);
return NULL;
}
tmp->next = glist;
glist = tmp;
tmp->fd = fd;
tmp->owner = NULL;
tmp->needsgain = 1;
tmp->fr.data = tmp->buf;
tmp->fr.frametype = AST_FRAME_VOICE;
@@ -332,7 +325,6 @@ static struct ast_filestream *wav_open(int fd)
/* datalen will vary for each frame */
tmp->fr.src = name;
tmp->fr.mallocd = 0;
tmp->lasttimeout = -1;
tmp->bytes = 0;
glistcnt++;
ast_pthread_mutex_unlock(&wav_lock);
@@ -358,11 +350,7 @@ static struct ast_filestream *wav_rewrite(int fd, char *comment)
free(tmp);
return NULL;
}
tmp->next = glist;
glist = tmp;
tmp->fd = fd;
tmp->owner = NULL;
tmp->lasttimeout = -1;
glistcnt++;
ast_pthread_mutex_unlock(&wav_lock);
ast_update_use_count();
@@ -371,60 +359,29 @@ static struct ast_filestream *wav_rewrite(int fd, char *comment)
return tmp;
}
static struct ast_frame *wav_read(struct ast_filestream *s)
{
return NULL;
}
static void wav_close(struct ast_filestream *s)
{
struct ast_filestream *tmp, *tmpl = NULL;
char zero = 0;
if (ast_pthread_mutex_lock(&wav_lock)) {
ast_log(LOG_WARNING, "Unable to lock wav 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(&wav_lock);
ast_update_use_count();
if (!tmp)
ast_log(LOG_WARNING, "Freeing a filestream we don't seem to own\n");
/* Pad to even length */
if (s->bytes & 0x1)
write(s->fd, &zero, 1);
close(s->fd);
free(s);
s = NULL;
#if 0
printf("bytes = %d\n", s->bytes);
#endif
}
static int ast_read_callback(void *data)
static struct ast_frame *wav_read(struct ast_filestream *s, int *whennext)
{
int retval = 0;
int res;
int delay;
int x;
struct ast_filestream *s = data;
short tmp[sizeof(s->buf) / 2];
int bytes = sizeof(tmp);
off_t here;
@@ -440,8 +397,7 @@ static int ast_read_callback(void *data)
if (res) {
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
}
s->owner->streamid = -1;
return 0;
return NULL;
}
#if __BYTE_ORDER == __BIG_ENDIAN
@@ -472,36 +428,8 @@ static int ast_read_callback(void *data)
s->fr.data = s->buf;
s->fr.mallocd = 0;
s->fr.samples = delay;
delay /= 8;
/* Lastly, process the frame */
if (delay != s->lasttimeout) {
s->owner->streamid = ast_sched_add(s->owner->sched, delay, ast_read_callback, s);
s->lasttimeout = delay;
} else {
retval = -1;
}
if (ast_write(s->owner, &s->fr)) {
ast_log(LOG_WARNING, "Failed to write frame\n");
s->owner->streamid = -1;
return 0;
}
return retval;
}
static int wav_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 wav_play(struct ast_filestream *s)
{
ast_read_callback(s);
return 0;
*whennext = delay;
return &s->fr;
}
static int wav_write(struct ast_filestream *fs, struct ast_frame *f)
@@ -607,8 +535,6 @@ int load_module()
return ast_format_register(name, exts, AST_FORMAT_SLINEAR,
wav_open,
wav_rewrite,
wav_apply,
wav_play,
wav_write,
wav_seek,
wav_trunc,
@@ -622,20 +548,6 @@ int load_module()
int unload_module()
{
struct ast_filestream *tmp, *tmpl;
if (ast_pthread_mutex_lock(&wav_lock)) {
ast_log(LOG_WARNING, "Unable to lock wav 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(&wav_lock);
return ast_format_unregister(name);
}