Remove "format_ogg_opus: New format"

This reverts commit 40aa28131bc30b4516da2b20eb1a1e043920169c.

ASTERISK-26426 #close

Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5

1 files changed, 0 insertions, 229 deletions

diff --git a/formats/format_ogg_opus.c b/formats/format_ogg_opus.c
deleted file mode 100644
index f6a4c6c67..000000000
--- a/formats/format_ogg_opus.c
+++ /dev/null
@@ -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
-);