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	A recent change to app_voicemail made it such that the module now assumes that all format modules are available while processing voicemail configuration. However, when autoloading modules, it was possible that app_voicemail was loaded before the format modules. Since format modules don't depend on anything, set a module load priority on them to ensure that they get loaded first when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2. The fix for 1.4 and 1.6.0 will require a different approach since the module load priority functionality is not present in the module API. (issue #16412) Reported by: jiddings git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
		
			
				
	
	
		
			143 lines
		
	
	
		
			3.5 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			143 lines
		
	
	
		
			3.5 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
/*
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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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	s->fr.subclass.codec = AST_FORMAT_SLINEAR16;
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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.codec != AST_FORMAT_SLINEAR16) {
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		ast_log(LOG_WARNING, "Asked to write non-slinear16 frame (%s)!\n", ast_getformatname(f->subclass.codec));
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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 const struct ast_format slin_f = {
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	.name = "sln16",
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	.exts = "sln16",
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	.format = AST_FORMAT_SLINEAR16,
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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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	if (ast_format_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_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 = 10,
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);
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