Totally redo file formats

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1130 65c4cc65-6c06-0410-ace0-fbb531ad65f3

formats/format_wav_gsm.c

@@ -43,21 +43,16 @@ struct ast_filestream {
 	/* This is what a filestream means to us */
 	int fd; /* Descriptor */
 	int bytes;
-	struct ast_channel *owner;
 	struct ast_frame fr;				/* Frame information */
 	char waste[AST_FRIENDLY_OFFSET];	/* Buffer for sending frames, etc */
 	char empty;							/* Empty character */
 	unsigned char gsm[66];				/* Two Real GSM Frames */
 	int foffset;
 	int secondhalf;						/* Are we on the second half */
-	int lasttimeout;
 	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;
 
@@ -341,10 +336,7 @@ static struct ast_filestream *wav_open(int fd)
 			free(tmp);
 			return NULL;
 		}
-		tmp->next = glist;
-		glist = tmp;
 		tmp->fd = fd;
-		tmp->owner = NULL;
 		tmp->fr.data = tmp->gsm;
 		tmp->fr.frametype = AST_FRAME_VOICE;
 		tmp->fr.subclass = AST_FORMAT_GSM;
@@ -352,7 +344,6 @@ static struct ast_filestream *wav_open(int fd)
 		tmp->fr.src = name;
 		tmp->fr.mallocd = 0;
 		tmp->secondhalf = 0;
-		tmp->lasttimeout = -1;
 		glistcnt++;
 		ast_pthread_mutex_unlock(&wav_lock);
 		ast_update_use_count();
@@ -377,11 +368,7 @@ 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();
@@ -390,42 +377,11 @@ 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);
@@ -434,14 +390,10 @@ static void wav_close(struct ast_filestream *s)
 	s = NULL;
 }
 
-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 = 20;
-	struct ast_filestream *s = data;
 	char msdata[66];
-	struct timeval tv;
 	/* Send a frame from the file to the appropriate channel */
 
 	s->fr.frametype = AST_FRAME_VOICE;
@@ -457,62 +409,15 @@ static int ast_read_callback(void *data)
 		if ((res = read(s->fd, msdata, 65)) != 65) {
 			if (res && (res != 1))
 				ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
-			s->owner->streamid = -1;
-			return 0;
+			return NULL;
 		}
 		/* Convert from MS format to two real GSM frames */
 		conv65(msdata, s->gsm);
 		s->fr.data = s->gsm;
 	}
 	s->secondhalf = !s->secondhalf;
-	/* 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;
-	}
-	if (s->last.tv_usec || s->last.tv_usec) {
-		int ms;
-		gettimeofday(&tv, NULL);
-		ms = 1000 * (tv.tv_sec - s->last.tv_sec) + 
-			(tv.tv_usec - s->last.tv_usec) / 1000;
-		s->last.tv_sec = tv.tv_sec;
-		s->last.tv_usec = tv.tv_usec;
-		if ((ms - delay) * (ms - delay) > 4) {
-			/* Compensate if we're more than 2 ms off */
-			s->adj -= (ms - delay);
-		}
-#if 0
-		fprintf(stdout, "Delay is %d, adjustment is %d, last was %d\n", delay, s->adj, ms);
-#endif
-		delay += s->adj;
-		if (delay < 1)
-			delay = 1;
-	} else
-		gettimeofday(&s->last, NULL);
-	if (s->lasttimeout != delay) {
-		/* We'll install the next timeout now. */
-		s->owner->streamid = ast_sched_add(s->owner->sched,
-				delay, ast_read_callback, s); 
-		s->lasttimeout = delay;
-	} else {
-		/* Just come back again at the same time */
-		retval = -1;
-	}
-	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 = 160;
+	return &s->fr;
 }
 
 static int wav_write(struct ast_filestream *fs, struct ast_frame *f)
@@ -609,8 +514,6 @@ int load_module()
 	return ast_format_register(name, exts, AST_FORMAT_GSM,
 								wav_open,
 								wav_rewrite,
-								wav_apply,
-								wav_play,
 								wav_write,
 								wav_seek,
 								wav_trunc,
@@ -624,20 +527,6 @@ 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);
 }