mirror of https://github.com/asterisk/asterisk
the Asterisk file format and codec translation modules (issue #6062) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12962 65c4cc65-6c06-0410-ace0-fbb531ad65f31.4
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2005, Digium, Inc.
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*
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* redice li <redice_li@yahoo.com>
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* Russell Bryant <russell@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 file format conversion CLI command using Asterisk formats and translators
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*
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* \author redice li <redice_li@yahoo.com>
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* \author Russell Bryant <russell@digium.com>
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*
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*/
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#include <stdio.h>
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#include <string.h>
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#include <stdlib.h>
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/channel.h"
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#include "asterisk/logger.h"
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#include "asterisk/module.h"
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#include "asterisk/cli.h"
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#include "asterisk/file.h"
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STANDARD_USECOUNT_DECL;
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static char *desc = "File format conversion CLI command";
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/*! \brief Split the filename to basename and extension */
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static int split_ext(char *filename, char **name, char **ext)
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{
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*name = *ext = filename;
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strsep(ext, ".");
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if (ast_strlen_zero(*name) || ast_strlen_zero(*ext))
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return -1;
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return 0;
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}
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/*! \brief Convert a file from one format to another */
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static int cli_audio_convert(int fd, int argc, char *argv[])
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{
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int ret = RESULT_FAILURE;
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struct ast_filestream *fs_in = NULL, *fs_out = NULL;
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struct ast_frame *f;
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struct timeval start;
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int cost;
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char *file_in = NULL, *file_out = NULL;
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char *name_in, *ext_in, *name_out, *ext_out;
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STANDARD_INCREMENT_USECOUNT;
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if (argc != 3 || ast_strlen_zero(argv[1]) || ast_strlen_zero(argv[2])) {
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ret = RESULT_SHOWUSAGE;
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goto fail_out;
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}
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if (!(file_in = ast_strdupa(argv[1])) || !(file_out = ast_strdupa(argv[2])))
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goto fail_out;
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if (split_ext(file_in, &name_in, &ext_in)) {
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ast_cli(fd, "'%s' is an invalid filename!\n", argv[1]);
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goto fail_out;
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}
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if (!(fs_in = ast_readfile(name_in, ext_in, NULL, O_RDONLY, 0, 0))) {
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ast_cli(fd, "Unable to open input file: %s\n", argv[1]);
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goto fail_out;
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}
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if (split_ext(file_out, &name_out, &ext_out)) {
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ast_cli(fd, "'%s' is an invalid filename!\n", argv[2]);
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goto fail_out;
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}
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if (!(fs_out = ast_writefile(name_out, ext_out, NULL, O_CREAT|O_TRUNC|O_WRONLY, 0, 0644))) {
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ast_cli(fd, "Unable to open output file: %s\n", argv[2]);
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goto fail_out;
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}
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start = ast_tvnow();
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while ((f = ast_readframe(fs_in))) {
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if (ast_writestream(fs_out, f)) {
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ast_cli(fd, "Failed to convert %s.%s to %s.%s!\n", name_in, ext_in, name_out, ext_out);
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goto fail_out;
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}
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}
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cost = ast_tvdiff_ms(ast_tvnow(), start);
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ast_cli(fd, "Converted %s.%s to %s.%s in %dms\n", name_in, ext_in, name_out, ext_out, cost);
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ret = RESULT_SUCCESS;
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fail_out:
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if (fs_out) {
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ast_closestream(fs_out);
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if (ret != RESULT_SUCCESS)
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ast_filedelete(name_out, ext_out);
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}
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if (fs_in)
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ast_closestream(fs_in);
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STANDARD_DECREMENT_USECOUNT;
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return ret;
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}
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static char usage_audio_convert[] =
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"Usage: convert <file_in> <file_out>\n"
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" Convert from file_in to file_out. If an absolute path is not given, the\n"
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"default Asterisk sounds directory will be used.\n\n"
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"Example:\n"
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" convert tt-weasels.gsm tt-weasels.ulaw\n";
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static struct ast_cli_entry audio_convert_cli={
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{ "convert" , NULL }, cli_audio_convert, "Convert audio files", usage_audio_convert
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};
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int unload_module(void)
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{
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return ast_cli_unregister(&audio_convert_cli);
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}
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int load_module(void)
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{
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return ast_cli_register(&audio_convert_cli);
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}
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char *description(void)
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{
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return desc;
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}
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int usecount(void)
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{
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int res;
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STANDARD_USECOUNT(res);
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return res;
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}
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char *key()
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{
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return ASTERISK_GPL_KEY;
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}
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