diff options
Diffstat (limited to 'formats/format_ogg_opus.c')
| -rw-r--r-- | formats/format_ogg_opus.c | 229 |
1 files changed, 229 insertions, 0 deletions
diff --git a/formats/format_ogg_opus.c b/formats/format_ogg_opus.c new file mode 100644 index 000000000..f6a4c6c67 --- /dev/null +++ b/formats/format_ogg_opus.c @@ -0,0 +1,229 @@ +/* + * 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 +); |