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asterisk/formats/format_sln.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Anthony Minessale
* Anthony Minessale (anthmct@yahoo.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.
*/
/*! \file
*
* \brief RAW SLINEAR Format
* \arg File name extensions: sln, raw
* \ingroup formats
*/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/mod_format.h"
#include "asterisk/module.h"
#include "asterisk/endian.h"
#define BUF_SIZE 320 /* 320 bytes, 160 samples */
#define SLIN_SAMPLES 160
static struct ast_frame *slinear_read(struct ast_filestream *s, int *whennext)
{
int res;
/* Send a frame from the file to the appropriate channel */
s->fr.frametype = AST_FRAME_VOICE;
s->fr.subclass = AST_FORMAT_SLINEAR;
s->fr.mallocd = 0;
AST_FRAME_SET_BUFFER(&s->fr, s->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
if ((res = fread(s->fr.data, 1, s->fr.datalen, s->f)) < 1) {
if (res)
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
return NULL;
}
*whennext = s->fr.samples = res/2;
s->fr.datalen = res;
return &s->fr;
}
static int slinear_write(struct ast_filestream *fs, struct ast_frame *f)
{
int res;
if (f->frametype != AST_FRAME_VOICE) {
ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
return -1;
}
if (f->subclass != AST_FORMAT_SLINEAR) {
ast_log(LOG_WARNING, "Asked to write non-slinear frame (%d)!\n", f->subclass);
return -1;
}
if ((res = fwrite(f->data, 1, f->datalen, fs->f)) != f->datalen) {
ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
return -1;
}
return 0;
}
static int slinear_seek(struct ast_filestream *fs, off_t sample_offset, int whence)
{
off_t offset=0,min,cur,max;
min = 0;
sample_offset <<= 1;
cur = ftello(fs->f);
fseeko(fs->f, 0, SEEK_END);
max = ftello(fs->f);
if (whence == SEEK_SET)
offset = sample_offset;
else if (whence == SEEK_CUR || whence == SEEK_FORCECUR)
offset = sample_offset + cur;
else if (whence == SEEK_END)
offset = max - sample_offset;
if (whence != SEEK_FORCECUR) {
offset = (offset > max)?max:offset;
}
/* always protect against seeking past begining. */
offset = (offset < min)?min:offset;
return fseeko(fs->f, offset, SEEK_SET);
}
static int slinear_trunc(struct ast_filestream *fs)
{
return ftruncate(fileno(fs->f), ftello(fs->f));
}
static off_t slinear_tell(struct ast_filestream *fs)
{
return ftello(fs->f) / 2;
}
static const struct ast_format slin_f = {
.name = "sln",
.exts = "sln|raw",
.format = AST_FORMAT_SLINEAR,
.write = slinear_write,
.seek = slinear_seek,
.trunc = slinear_trunc,
.tell = slinear_tell,
.read = slinear_read,
.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
};
static int load_module(void)
{
More changes to change return values from load_module functions. (issue #11096) Patches: codec_adpcm.c.patch uploaded by moy (license 222) codec_alaw.c.patch uploaded by moy (license 222) codec_a_mu.c.patch uploaded by moy (license 222) codec_g722.c.patch uploaded by moy (license 222) codec_g726.c.diff uploaded by moy (license 222) codec_gsm.c.patch uploaded by moy (license 222) codec_ilbc.c.patch uploaded by moy (license 222) codec_lpc10.c.patch uploaded by moy (license 222) codec_speex.c.patch uploaded by moy (license 222) codec_ulaw.c.patch uploaded by moy (license 222) codec_zap.c.patch uploaded by moy (license 222) format_g723.c.patch uploaded by moy (license 222) format_g726.c.patch uploaded by moy (license 222) format_g729.c.patch uploaded by moy (license 222) format_gsm.c.patch uploaded by moy (license 222) format_h263.c.patch uploaded by moy (license 222) format_h264.c.patch uploaded by moy (license 222) format_ilbc.c.patch uploaded by moy (license 222) format_jpeg.c.patch uploaded by moy (license 222) format_ogg_vorbis.c.patch uploaded by moy (license 222) format_pcm.c.patch uploaded by moy (license 222) format_sln.c.patch uploaded by moy (license 222) format_vox.c.patch uploaded by moy (license 222) format_wav.c.patch uploaded by moy (license 222) format_wav_gsm.c.patch uploaded by moy (license 222) res_adsi.c.patch uploaded by eliel (license 64) res_ael_share.c.patch uploaded by eliel (license 64) res_clioriginate.c.patch uploaded by eliel (license 64) res_convert.c.patch uploaded by eliel (license 64) res_indications.c.patch uploaded by eliel (license 64) res_musiconhold.c.patch uploaded by eliel (license 64) res_smdi.c.patch uploaded by eliel (license 64) res_speech.c.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87889 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-31 19:24:29 +00:00
if (ast_format_register(&slin_f))
return AST_MODULE_LOAD_FAILURE;
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
return ast_format_unregister(slin_f.name);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Raw Signed Linear Audio support (SLN)");