mirror of https://github.com/asterisk/asterisk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327137 65c4cc65-6c06-0410-ace0-fbb531ad65f310-digiumphones
David Vossel
14 years ago
parent
a86c1d68e9
commit
331759b64c
@ -1,142 +0,0 @@
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2008, Anthony Minessale and Digium, Inc.
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* Anthony Minessale (anthmct@yahoo.com)
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* Kevin P. Fleming <kpfleming@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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/*! \file
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*
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* \brief RAW SLINEAR 16 Format
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* \arg File name extensions: sln16
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* \ingroup formats
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*/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/mod_format.h"
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#include "asterisk/module.h"
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#include "asterisk/endian.h"
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#define BUF_SIZE 640 /* 640 bytes, 320 samples */
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#define SLIN_SAMPLES 320
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static struct ast_frame *slinear_read(struct ast_filestream *s, int *whennext)
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{
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int res;
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/* Send a frame from the file to the appropriate channel */
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s->fr.frametype = AST_FRAME_VOICE;
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ast_format_set(&s->fr.subclass.format, AST_FORMAT_SLINEAR16, 0);
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s->fr.mallocd = 0;
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AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
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if ((res = fread(s->fr.data.ptr, 1, s->fr.datalen, s->f)) < 1) {
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if (res)
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ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
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return NULL;
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}
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*whennext = s->fr.samples = res/2;
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s->fr.datalen = res;
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return &s->fr;
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}
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static int slinear_write(struct ast_filestream *fs, struct ast_frame *f)
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{
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int res;
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if (f->frametype != AST_FRAME_VOICE) {
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ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
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return -1;
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}
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if (f->subclass.format.id != AST_FORMAT_SLINEAR16) {
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ast_log(LOG_WARNING, "Asked to write non-slinear16 frame (%s)!\n", ast_getformatname(&f->subclass.format));
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return -1;
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}
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if ((res = fwrite(f->data.ptr, 1, f->datalen, fs->f)) != f->datalen) {
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ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
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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 slinear_seek(struct ast_filestream *fs, off_t sample_offset, int whence)
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{
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off_t offset = 0, min = 0, cur, max;
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sample_offset <<= 1;
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cur = ftello(fs->f);
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fseeko(fs->f, 0, SEEK_END);
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max = ftello(fs->f);
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if (whence == SEEK_SET)
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offset = sample_offset;
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else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
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offset = sample_offset + cur;
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else if (whence == SEEK_END)
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offset = max - sample_offset;
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if (whence != SEEK_FORCECUR)
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offset = (offset > max) ? max : offset;
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/* always protect against seeking past begining. */
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offset = (offset < min) ? min : offset;
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return fseeko(fs->f, offset, SEEK_SET);
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}
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static int slinear_trunc(struct ast_filestream *fs)
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{
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return ftruncate(fileno(fs->f), ftello(fs->f));
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}
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static off_t slinear_tell(struct ast_filestream *fs)
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{
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return ftello(fs->f) / 2;
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}
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static struct ast_format_def slin_f = {
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.name = "sln16",
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.exts = "sln16",
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.write = slinear_write,
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.seek = slinear_seek,
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.trunc = slinear_trunc,
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.tell = slinear_tell,
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.read = slinear_read,
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.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
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};
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static int load_module(void)
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{
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ast_format_set(&slin_f.format, AST_FORMAT_SLINEAR16, 0);
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if (ast_format_def_register(&slin_f))
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return AST_MODULE_LOAD_FAILURE;
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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(slin_f.name);
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Raw Signed Linear 16KHz Audio support (SLN16)",
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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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