mirror of https://github.com/asterisk/asterisk
George Joseph
9 years ago
committed by
Gerrit Code Review
commit
d205d8d4a1
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2016, Digium, Inc.
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*
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* Mark Michelson <mmichelson@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*** MODULEINFO
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<depend>opusfile</depend>
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include <opus/opus.h>
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#include <opus/opusfile.h>
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#include "asterisk/mod_format.h"
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#include "asterisk/utils.h"
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#include "asterisk/module.h"
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#include "asterisk/format_cache.h"
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/* 120ms of 48KHz audio */
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#define SAMPLES_MAX 5760
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#define BUF_SIZE (2 * SAMPLES_MAX)
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struct ogg_opus_desc {
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OggOpusFile *of;
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};
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static int fread_wrapper(void *_stream, unsigned char *_ptr, int _nbytes)
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{
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FILE *stream = _stream;
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size_t bytes_read;
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if (!stream || _nbytes < 0) {
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return -1;
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}
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bytes_read = fread(_ptr, 1, _nbytes, stream);
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return bytes_read > 0 || feof(stream) ? (int) bytes_read : OP_EREAD;
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}
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static int fseek_wrapper(void *_stream, opus_int64 _offset, int _whence)
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{
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FILE *stream = _stream;
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return fseeko(stream, (off_t) _offset, _whence);
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}
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static opus_int64 ftell_wrapper(void *_stream)
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{
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FILE *stream = _stream;
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return ftello(stream);
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}
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static int ogg_opus_open(struct ast_filestream *s)
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{
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struct ogg_opus_desc *desc = (struct ogg_opus_desc *) s->_private;
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OpusFileCallbacks cb = {
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.read = fread_wrapper,
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.seek = fseek_wrapper,
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.tell = ftell_wrapper,
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.close = NULL,
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};
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memset(desc, 0, sizeof(*desc));
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desc->of = op_open_callbacks(s->f, &cb, NULL, 0, NULL);
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if (!desc->of) {
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return -1;
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}
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return 0;
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}
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static int ogg_opus_rewrite(struct ast_filestream *s, const char *comment)
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{
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/* XXX Unimplemented. We currently only can read from OGG/Opus streams */
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ast_log(LOG_ERROR, "Cannot write OGG/Opus streams. Sorry :(\n");
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return -1;
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}
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static int ogg_opus_write(struct ast_filestream *fs, struct ast_frame *f)
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{
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/* XXX Unimplemented. We currently only can read from OGG/Opus streams */
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ast_log(LOG_ERROR, "Cannot write OGG/Opus streams. Sorry :(\n");
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return -1;
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}
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static int ogg_opus_seek(struct ast_filestream *fs, off_t sample_offset, int whence)
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{
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int seek_result = -1;
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off_t relative_pcm_pos;
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struct ogg_opus_desc *desc = fs->_private;
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switch (whence) {
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case SEEK_SET:
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seek_result = op_pcm_seek(desc->of, sample_offset);
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break;
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case SEEK_CUR:
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if ((relative_pcm_pos = op_pcm_tell(desc->of)) < 0) {
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seek_result = -1;
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break;
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}
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seek_result = op_pcm_seek(desc->of, relative_pcm_pos + sample_offset);
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break;
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case SEEK_END:
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if ((relative_pcm_pos = op_pcm_total(desc->of, -1)) < 0) {
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seek_result = -1;
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break;
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}
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seek_result = op_pcm_seek(desc->of, relative_pcm_pos - sample_offset);
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break;
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default:
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ast_log(LOG_WARNING, "Unknown *whence* to seek on OGG/Opus streams!\n");
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break;
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}
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/* normalize error value to -1,0 */
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return (seek_result == 0) ? 0 : -1;
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}
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static int ogg_opus_trunc(struct ast_filestream *fs)
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{
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/* XXX Unimplemented. This is only used when recording, and we don't support that right now. */
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ast_log(LOG_ERROR, "Truncation is not supported on OGG/Opus streams!\n");
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return -1;
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}
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static off_t ogg_opus_tell(struct ast_filestream *fs)
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{
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struct ogg_opus_desc *desc = fs->_private;
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off_t pos;
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pos = (off_t) op_pcm_tell(desc->of);
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if (pos < 0) {
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return -1;
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}
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return pos;
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}
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static struct ast_frame *ogg_opus_read(struct ast_filestream *fs, int *whennext)
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{
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struct ogg_opus_desc *desc = fs->_private;
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int hole = 1;
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int samples_read;
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opus_int16 *out_buf;
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AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
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out_buf = (opus_int16 *) fs->fr.data.ptr;
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while (hole) {
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samples_read = op_read(
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desc->of,
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out_buf,
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SAMPLES_MAX,
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NULL);
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if (samples_read != OP_HOLE) {
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hole = 0;
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}
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}
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if (samples_read <= 0) {
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return NULL;
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}
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fs->fr.datalen = samples_read * 2;
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fs->fr.samples = samples_read;
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*whennext = fs->fr.samples;
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return &fs->fr;
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}
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static void ogg_opus_close(struct ast_filestream *fs)
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{
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struct ogg_opus_desc *desc = fs->_private;
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op_free(desc->of);
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}
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static struct ast_format_def opus_f = {
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.name = "ogg_opus",
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.exts = "opus",
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.open = ogg_opus_open,
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.rewrite = ogg_opus_rewrite,
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.write = ogg_opus_write,
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.seek = ogg_opus_seek,
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.trunc = ogg_opus_trunc,
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.tell = ogg_opus_tell,
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.read = ogg_opus_read,
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.close = ogg_opus_close,
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.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
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.desc_size = sizeof(struct ogg_opus_desc),
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};
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static int load_module(void)
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{
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opus_f.format = ast_format_slin48;
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if (ast_format_def_register(&opus_f)) {
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return AST_MODULE_LOAD_FAILURE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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static int unload_module(void)
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{
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return ast_format_def_unregister(opus_f.name);
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "OGG/Opus audio",
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.support_level = AST_MODULE_SUPPORT_CORE,
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.load = load_module,
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.unload = unload_module,
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.load_pri = AST_MODPRI_APP_DEPEND
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);
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