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Remove "format_ogg_opus: New format"

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This reverts commit 40aa28131b.

ASTERISK-26426 #close

Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5
This commit is contained in:
Kevin Harwell
2016-09-29 14:02:37 -05:00
parent 2d2a8944be
commit d31ffb421c
6 changed files with 0 additions and 392 deletions
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229
formats/format_ogg_opus.c
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@@ -1,229 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2016, Digium, Inc.
*
* Mark Michelson <mmichelson@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.
*/
/*** MODULEINFO
<depend>opusfile</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <opus/opus.h>
#include <opus/opusfile.h>
#include "asterisk/mod_format.h"
#include "asterisk/utils.h"
#include "asterisk/module.h"
#include "asterisk/format_cache.h"
/* 120ms of 48KHz audio */
#define SAMPLES_MAX 5760
#define BUF_SIZE (2 * SAMPLES_MAX)
struct ogg_opus_desc {
OggOpusFile *of;
};
static int fread_wrapper(void *_stream, unsigned char *_ptr, int _nbytes)
{
FILE *stream = _stream;
size_t bytes_read;
if (!stream || _nbytes < 0) {
return -1;
}
bytes_read = fread(_ptr, 1, _nbytes, stream);
return bytes_read > 0 || feof(stream) ? (int) bytes_read : OP_EREAD;
}
static int fseek_wrapper(void *_stream, opus_int64 _offset, int _whence)
{
FILE *stream = _stream;
return fseeko(stream, (off_t) _offset, _whence);
}
static opus_int64 ftell_wrapper(void *_stream)
{
FILE *stream = _stream;
return ftello(stream);
}
static int ogg_opus_open(struct ast_filestream *s)
{
struct ogg_opus_desc *desc = (struct ogg_opus_desc *) s->_private;
OpusFileCallbacks cb = {
.read = fread_wrapper,
.seek = fseek_wrapper,
.tell = ftell_wrapper,
.close = NULL,
};
memset(desc, 0, sizeof(*desc));
desc->of = op_open_callbacks(s->f, &cb, NULL, 0, NULL);
if (!desc->of) {
return -1;
}
return 0;
}
static int ogg_opus_rewrite(struct ast_filestream *s, const char *comment)
{
/* XXX Unimplemented. We currently only can read from OGG/Opus streams */
ast_log(LOG_ERROR, "Cannot write OGG/Opus streams. Sorry :(\n");
return -1;
}
static int ogg_opus_write(struct ast_filestream *fs, struct ast_frame *f)
{
/* XXX Unimplemented. We currently only can read from OGG/Opus streams */
ast_log(LOG_ERROR, "Cannot write OGG/Opus streams. Sorry :(\n");
return -1;
}
static int ogg_opus_seek(struct ast_filestream *fs, off_t sample_offset, int whence)
{
int seek_result = -1;
off_t relative_pcm_pos;
struct ogg_opus_desc *desc = fs->_private;
switch (whence) {
case SEEK_SET:
seek_result = op_pcm_seek(desc->of, sample_offset);
break;
case SEEK_CUR:
if ((relative_pcm_pos = op_pcm_tell(desc->of)) < 0) {
seek_result = -1;
break;
}
seek_result = op_pcm_seek(desc->of, relative_pcm_pos + sample_offset);
break;
case SEEK_END:
if ((relative_pcm_pos = op_pcm_total(desc->of, -1)) < 0) {
seek_result = -1;
break;
}
seek_result = op_pcm_seek(desc->of, relative_pcm_pos - sample_offset);
break;
default:
ast_log(LOG_WARNING, "Unknown *whence* to seek on OGG/Opus streams!\n");
break;
}
/* normalize error value to -1,0 */
return (seek_result == 0) ? 0 : -1;
}
static int ogg_opus_trunc(struct ast_filestream *fs)
{
/* XXX Unimplemented. This is only used when recording, and we don't support that right now. */
ast_log(LOG_ERROR, "Truncation is not supported on OGG/Opus streams!\n");
return -1;
}
static off_t ogg_opus_tell(struct ast_filestream *fs)
{
struct ogg_opus_desc *desc = fs->_private;
off_t pos;
pos = (off_t) op_pcm_tell(desc->of);
if (pos < 0) {
return -1;
}
return pos;
}
static struct ast_frame *ogg_opus_read(struct ast_filestream *fs, int *whennext)
{
struct ogg_opus_desc *desc = fs->_private;
int hole = 1;
int samples_read;
opus_int16 *out_buf;
AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
out_buf = (opus_int16 *) fs->fr.data.ptr;
while (hole) {
samples_read = op_read(
desc->of,
out_buf,
SAMPLES_MAX,
NULL);
if (samples_read != OP_HOLE) {
hole = 0;
}
}
if (samples_read <= 0) {
return NULL;
}
fs->fr.datalen = samples_read * 2;
fs->fr.samples = samples_read;
*whennext = fs->fr.samples;
return &fs->fr;
}
static void ogg_opus_close(struct ast_filestream *fs)
{
struct ogg_opus_desc *desc = fs->_private;
op_free(desc->of);
}
static struct ast_format_def opus_f = {
.name = "ogg_opus",
.exts = "opus",
.open = ogg_opus_open,
.rewrite = ogg_opus_rewrite,
.write = ogg_opus_write,
.seek = ogg_opus_seek,
.trunc = ogg_opus_trunc,
.tell = ogg_opus_tell,
.read = ogg_opus_read,
.close = ogg_opus_close,
.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
.desc_size = sizeof(struct ogg_opus_desc),
};
static int load_module(void)
{
opus_f.format = ast_format_slin48;
if (ast_format_def_register(&opus_f)) {
return AST_MODULE_LOAD_FAILURE;
}
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
return ast_format_def_unregister(opus_f.name);
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "OGG/Opus audio",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND
);