add a file-format driver for 16KHz signed linear... which may or may not work

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96862 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 134 insertions, 0 deletions

diff --git a/formats/format_sln16.c b/formats/format_sln16.c
new file mode 100644
index 000000000..ab01d80dd
--- /dev/null
+++ b/formats/format_sln16.c
@@ -0,0 +1,134 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2008, Anthony Minessale and Digium, Inc.
+ * Anthony Minessale (anthmct@yahoo.com)
+ * Kevin P. Fleming <kpfleming@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief RAW SLINEAR 16 Format
+ * \arg File name extensions: sln16
+ * \ingroup formats
+ */
+ 
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/mod_format.h"
+#include "asterisk/module.h"
+#include "asterisk/endian.h"
+
+#define BUF_SIZE	640		/* 640 bytes, 320 samples */
+#define	SLIN_SAMPLES	320
+
+static struct ast_frame *slinear_read(struct ast_filestream *s, int *whennext)
+{
+	int res;
+	/* Send a frame from the file to the appropriate channel */
+
+	s->fr.frametype = AST_FRAME_VOICE;
+	s->fr.subclass = AST_FORMAT_SLINEAR16;
+	s->fr.mallocd = 0;
+	AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
+	if ((res = fread(s->fr.data, 1, s->fr.datalen, s->f)) < 1) {
+		if (res)
+			ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
+		return NULL;
+	}
+	*whennext = s->fr.samples = res/2;
+	s->fr.datalen = res;
+	return &s->fr;
+}
+
+static int slinear_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_SLINEAR16) {
+		ast_log(LOG_WARNING, "Asked to write non-slinear16 frame (%d)!\n", f->subclass);
+		return -1;
+	}
+	if ((res = fwrite(f->data, 1, f->datalen, fs->f)) != f->datalen) {
+			ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
+			return -1;
+	}
+	return 0;
+}
+
+static int slinear_seek(struct ast_filestream *fs, off_t sample_offset, int whence)
+{
+	off_t offset=0, min, cur, max;
+
+	min = 0;
+	sample_offset <<= 1;
+	cur = ftello(fs->f);
+	fseeko(fs->f, 0, SEEK_END);
+	max = ftello(fs->f);
+	if (whence == SEEK_SET)
+		offset = sample_offset;
+	else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
+		offset = sample_offset + cur;
+	else if (whence == SEEK_END)
+		offset = max - sample_offset;
+	if (whence != SEEK_FORCECUR) {
+		offset = (offset > max)?max:offset;
+	}
+	/* always protect against seeking past begining. */
+	offset = (offset < min)?min:offset;
+
+	return fseeko(fs->f, offset, SEEK_SET);
+}
+
+static int slinear_trunc(struct ast_filestream *fs)
+{
+	return ftruncate(fileno(fs->f), ftello(fs->f));
+}
+
+static off_t slinear_tell(struct ast_filestream *fs)
+{
+	return ftello(fs->f) / 2;
+}
+
+static const struct ast_format slin_f = {
+	.name = "sln16",
+	.exts = "sln16",
+	.format = AST_FORMAT_SLINEAR16,
+	.write = slinear_write,
+	.seek = slinear_seek,
+	.trunc = slinear_trunc,
+	.tell = slinear_tell,
+	.read = slinear_read,
+	.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
+};
+
+static int load_module(void)
+{
+	if (ast_format_register(&slin_f))
+		return AST_MODULE_LOAD_FAILURE;
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	return ast_format_unregister(slin_f.name);
+}	
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Raw Signed Linear 16KHz Audio support (SLN16)");