Version 0.1.0 from FTP

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 355 insertions, 0 deletions

diff --git a/formats/format_wav.c b/formats/format_wav.c
new file mode 100755
index 000000000..e3265d379
--- /dev/null
+++ b/formats/format_wav.c
@@ -0,0 +1,355 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Microsoft WAV File Format using libaudiofile 
+ * 
+ * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/channel.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/sched.h>
+#include <asterisk/module.h>
+#include <arpa/inet.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <unistd.h>
+#include <errno.h>
+#include <string.h>
+#include <pthread.h>
+#include <audiofile.h>
+
+
+/* Read 320 samples at a time, max */ 
+#define WAV_MAX_SIZE 320
+
+/* Fudge in milliseconds */
+#define WAV_FUDGE 2
+
+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 */
+	/* Audio File */
+	AFfilesetup afs;
+	AFfilehandle af;
+	int lasttimeout;
+	struct ast_channel *owner;
+	struct ast_filestream *next;
+	struct ast_frame fr;				/* Frame information */
+	char waste[AST_FRIENDLY_OFFSET];	/* Buffer for sending frames, etc */
+	short samples[WAV_MAX_SIZE];
+};
+
+
+static struct ast_filestream *glist = NULL;
+static pthread_mutex_t wav_lock = PTHREAD_MUTEX_INITIALIZER;
+static int glistcnt = 0;
+
+static char *name = "wav";
+static char *desc = "Microsoft WAV format (PCM/16, 8000Hz mono)";
+static char *exts = "wav";
+
+static struct ast_filestream *wav_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;
+	int notok = 0;
+	int fmt, width;
+	double rate;
+	if ((tmp = malloc(sizeof(struct ast_filestream)))) {
+		tmp->afs = afNewFileSetup();
+		if (!tmp->afs) {
+			ast_log(LOG_WARNING, "Unable to create file setup\n");
+			free(tmp);
+			return NULL;
+		}
+		afInitFileFormat(tmp->afs, AF_FILE_WAVE);
+		tmp->af = afOpenFD(fd, "r", tmp->afs);
+		if (!tmp->af) {
+			afFreeFileSetup(tmp->afs);
+			ast_log(LOG_WARNING, "Unable to open file descriptor\n");
+			free(tmp);
+			return NULL;
+		}
+#if 0
+		afGetFileFormat(tmp->af, &version);
+		if (version != AF_FILE_WAVE) {
+			ast_log(LOG_WARNING, "This is not a wave file (%d)\n", version);
+			notok++;
+		}
+#endif
+		/* Read the format and make sure it's exactly what we seek. */
+		if (afGetChannels(tmp->af, AF_DEFAULT_TRACK) != 1) {
+			ast_log(LOG_WARNING, "Invalid number of channels %d.  Should be mono (1)\n", afGetChannels(tmp->af, AF_DEFAULT_TRACK));
+			notok++;
+		}
+		afGetSampleFormat(tmp->af, AF_DEFAULT_TRACK, &fmt, &width);
+		if (fmt != AF_SAMPFMT_TWOSCOMP) {
+			ast_log(LOG_WARNING, "Input file is not signed\n");
+			notok++;
+		}
+		rate = afGetRate(tmp->af, AF_DEFAULT_TRACK);
+		if ((rate < 7900) || (rate > 8100)) {
+			ast_log(LOG_WARNING, "Rate %f is not close enough to 8000 Hz\n", rate);
+			notok++;
+		}
+		if (width != 16) {
+			ast_log(LOG_WARNING, "Input file is not 16-bit\n");
+			notok++;
+		}
+		if (notok) {
+			afCloseFile(tmp->af);
+			afFreeFileSetup(tmp->afs);
+			free(tmp);
+			return NULL;
+		}
+		if (pthread_mutex_lock(&wav_lock)) {
+			afCloseFile(tmp->af);
+			afFreeFileSetup(tmp->afs);
+			ast_log(LOG_WARNING, "Unable to lock wav list\n");
+			free(tmp);
+			return NULL;
+		}
+		tmp->next = glist;
+		glist = tmp;
+		tmp->fd = fd;
+		tmp->owner = NULL;
+		tmp->fr.data = tmp->samples;
+		tmp->fr.frametype = AST_FRAME_VOICE;
+		tmp->fr.subclass = AST_FORMAT_SLINEAR;
+		/* datalen will vary for each frame */
+		tmp->fr.src = name;
+		tmp->fr.mallocd = 0;
+		tmp->lasttimeout = -1;
+		glistcnt++;
+		pthread_mutex_unlock(&wav_lock);
+		ast_update_use_count();
+	}
+	return tmp;
+}
+
+static struct ast_filestream *wav_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)))) {
+		tmp->afs = afNewFileSetup();
+		if (!tmp->afs) {
+			ast_log(LOG_WARNING, "Unable to create file setup\n");
+			free(tmp);
+			return NULL;
+		}
+		/* WAV format */
+		afInitFileFormat(tmp->afs, AF_FILE_WAVE);
+		/* Mono */
+		afInitChannels(tmp->afs, AF_DEFAULT_TRACK, 1);
+		/* Signed linear, 16-bit */
+		afInitSampleFormat(tmp->afs, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+		/* 8000 Hz */
+		afInitRate(tmp->afs, AF_DEFAULT_TRACK, (double)8000.0);
+		tmp->af = afOpenFD(fd, "w", tmp->afs);
+		if (!tmp->af) {
+			afFreeFileSetup(tmp->afs);
+			ast_log(LOG_WARNING, "Unable to open file descriptor\n");
+			free(tmp);
+			return NULL;
+		}
+		if (pthread_mutex_lock(&wav_lock)) {
+			ast_log(LOG_WARNING, "Unable to lock wav list\n");
+			free(tmp);
+			return NULL;
+		}
+		tmp->next = glist;
+		glist = tmp;
+		tmp->fd = fd;
+		tmp->owner = NULL;
+		tmp->lasttimeout = -1;
+		glistcnt++;
+		pthread_mutex_unlock(&wav_lock);
+		ast_update_use_count();
+	} else
+		ast_log(LOG_WARNING, "Out of memory\n");
+	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;
+	if (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;
+	}
+	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");
+	afCloseFile(tmp->af);
+	afFreeFileSetup(tmp->afs);
+	close(s->fd);
+	free(s);
+}
+
+static int ast_read_callback(void *data)
+{
+	u_int32_t delay = -1;
+	int retval = 0;
+	int res;
+	struct ast_filestream *s = data;
+	/* Send a frame from the file to the appropriate channel */
+
+	if ((res = afReadFrames(s->af, AF_DEFAULT_TRACK, s->samples, sizeof(s->samples)/2)) < 1) {
+		if (res)
+			ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
+		s->owner->streamid = -1;
+		return 0;
+	}
+	/* Per 8 samples, one milisecond */
+	delay = res / 8;
+	s->fr.frametype = AST_FRAME_VOICE;
+	s->fr.subclass = AST_FORMAT_SLINEAR;
+	s->fr.offset = AST_FRIENDLY_OFFSET;
+	s->fr.datalen = res * 2;
+	s->fr.data = s->samples;
+	s->fr.mallocd = 0;
+	s->fr.timelen = delay;
+	/* Unless there is no delay, we're going to exit out as soon as we
+	   have processed the current frame. */
+	/* If there is a delay, lets schedule the next event */
+	if (delay != s->lasttimeout) {
+		/* 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;
+	}
+	/* 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;
+	}
+	
+	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;
+	ast_read_callback(s);
+	return 0;
+}
+
+static int wav_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_SLINEAR) {
+		ast_log(LOG_WARNING, "Asked to write non-signed linear frame (%d)!\n", f->subclass);
+		return -1;
+	}
+	if ((res = afWriteFrames(fs->af, AF_DEFAULT_TRACK, f->data, f->datalen/2)) != f->datalen/2) {
+		ast_log(LOG_WARNING, "Unable to write frame: res=%d (%s)\n", res, strerror(errno));
+		return -1;
+	}	
+	return 0;
+}
+
+char *wav_getcomment(struct ast_filestream *s)
+{
+	return NULL;
+}
+
+int load_module()
+{
+	return ast_format_register(name, exts, AST_FORMAT_SLINEAR,
+								wav_open,
+								wav_rewrite,
+								wav_apply,
+								wav_write,
+								wav_read,
+								wav_close,
+								wav_getcomment);								
+								
+								
+}
+
+int unload_module()
+{
+	struct ast_filestream *tmp, *tmpl;
+	if (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);
+		tmpl = tmp;
+		tmp = tmp->next;
+		free(tmpl);
+	}
+	pthread_mutex_unlock(&wav_lock);
+	return ast_format_unregister(name);
+}	
+
+int usecount()
+{
+	int res;
+	if (pthread_mutex_lock(&wav_lock)) {
+		ast_log(LOG_WARNING, "Unable to lock wav list\n");
+		return -1;
+	}
+	res = glistcnt;
+	pthread_mutex_unlock(&wav_lock);
+	return res;
+}
+
+char *description()
+{
+	return desc;
+}
+