libmad是一个开源mp3解码库，其对mp3解码算法做了很多优化，性能较好，很多播放器如mplayer、xmms等都是使用这个开源库进行解码的；如果要设计mp3播放器而又不想研 究mp3解码算法的话，libmad是个不错的选择。关于该库的使用，叙述如下。
一：安装Libmad:
1、在网上下载：Libmad库的使用.pdf文档和libmad-0.15.lb.tar.gz压缩包（ http://down.51cto.com/data/1087041（免费下载））
2、解压：tar -zxvf libmad-0.15.lb.tar.gz
以下过程在Readme和INSTALL文件中列了出来，应学会自己看选项进行操作：
3、cd libmad-0.15.lb
4、./configure
5、make
6、make check
7、make install
（若最后有错误信息，说明你用的gcc版本太高，该版本的gcc有"-fforce-mem"参数，打开根目录下的Makefile去掉里面的"-fforce-mem"就OK了。）
结果：产生一个 .libs 目录
--------------------------------------------------------------------------------------------------------------————————————————
然后按照Libmad库的使用.pdf文档中的提示继续往下进行。
二：查看示例代码 minimad.c:
minimad.c
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/*
* libmad - MPEG audio decoder library
* Copyright (C) 2000-2004 Underbit Technologies, Inc.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
* $Id: minimad.c,v 1.4 2004/01/23 09:41:32 rob Exp $
*/
# include <stdio.h>
# include <unistd.h>
# include <sys/stat.h>
# include <sys/mman.h>
# include "mad.h"
/*
* This is perhaps the simplest example use of the MAD high-level API.
* Standard input is mapped into memory via mmap(), then the high-level API
* is invoked with three callbacks: input, output, and error. The output
* callback converts MAD's high-resolution PCM samples to 16 bits, then
* writes them to standard output in little-endian, stereo-interleaved
* format.
*/
static
int
decode(unsigned
char
const
*, unsigned
long
);
int
main(
int
argc,
char
*argv[])
{
struct
stat stat;
void
*fdm;
if
(argc != 1)
return
1;
if
(fstat(STDIN_FILENO, &stat) == -1 ||
stat.st_size == 0)
return
2;
fdm = mmap(0, stat.st_size, PROT_READ, MAP_SHARED, STDIN_FILENO, 0);
if
(fdm == MAP_FAILED)
return
3;
decode(fdm, stat.st_size);
if
(munmap(fdm, stat.st_size) == -1)
return
4;
return
0;
}
/*
* This is a private message structure. A generic pointer to this structure
* is passed to each of the callback functions. Put here any data you need
* to access from within the callbacks.
*/
struct
buffer {
unsigned
char
const
*start;
unsigned
long
length;
};
/*
* This is the input callback. The purpose of this callback is to (re)fill
* the stream buffer which is to be decoded. In this example, an entire file
* has been mapped into memory, so we just call mad_stream_buffer() with the
* address and length of the mapping. When this callback is called a second
* time, we are finished decoding.
*/
static
enum
mad_flow input(
void
*data,
struct
mad_stream *stream)
{
struct
buffer *buffer = data;
if
(!buffer->length)
return
MAD_FLOW_STOP;
mad_stream_buffer(stream, buffer->start, buffer->length);
buffer->length = 0;
return
MAD_FLOW_CONTINUE;
}
/*
* The following utility routine performs simple rounding, clipping, and
* scaling of MAD's high-resolution samples down to 16 bits. It does not
* perform any dithering or noise shaping, which would be recommended to
* obtain any exceptional audio quality. It is therefore not recommended to
* use this routine if high-quality output is desired.
*/
static
inline
signed
int
scale(mad_fixed_t sample)
{
/* round */
sample += (1L << (MAD_F_FRACBITS - 16));
/* clip */
if
(sample >= MAD_F_ONE)
sample = MAD_F_ONE - 1;
else
if
(sample < -MAD_F_ONE)
sample = -MAD_F_ONE;
/* quantize */
return
sample >> (MAD_F_FRACBITS + 1 - 16);
}
/*
* This is the output callback function. It is called after each frame of
* MPEG audio data has been completely decoded. The purpose of this callback
* is to output (or play) the decoded PCM audio.
*/
static
enum
mad_flow output(
void
*data,
struct
mad_header
const
*header,
struct
mad_pcm *pcm)
{
unsigned
int
nchannels, nsamples;
mad_fixed_t
const
*left_ch, *right_ch;
/* pcm->samplerate contains the sampling frequency */
nchannels = pcm->channels;
nsamples = pcm->length;
left_ch = pcm->samples[0];
right_ch = pcm->samples[1];
while
(nsamples--) {
signed
int
sample;
/* output sample(s) in 16-bit signed little-endian PCM */
sample = scale(*left_ch++);
putchar
((sample >> 0) & 0xff);
putchar
((sample >> 8) & 0xff);
if
(nchannels == 2) {
sample = scale(*right_ch++);
putchar
((sample >> 0) & 0xff);
putchar
((sample >> 8) & 0xff);
}
}
return
MAD_FLOW_CONTINUE;
}
/*
* This is the error callback function. It is called whenever a decoding
* error occurs. The error is indicated by stream->error; the list of
* possible MAD_ERROR_* errors can be found in the mad.h (or stream.h)
* header file.
*/
static
enum
mad_flow error(
void
*data,
struct
mad_stream *stream,
struct
mad_frame *frame)
{
struct
buffer *buffer = data;
fprintf
(stderr,
"decoding error 0x%04x (%s) at byte offset %u\n"
,
stream->error, mad_stream_errorstr(stream),
stream->this_frame - buffer->start);
/* return MAD_FLOW_BREAK here to stop decoding (and propagate an error) */
return
MAD_FLOW_CONTINUE;
}
/*
* This is the function called by main() above to perform all the decoding.
* It instantiates a decoder object and configures it with the input,
* output, and error callback functions above. A single call to
* mad_decoder_run() continues until a callback function returns
* MAD_FLOW_STOP (to stop decoding) or MAD_FLOW_BREAK (to stop decoding and
* signal an error).
*/
static
int
decode(unsigned
char
const
*start, unsigned
long
length)
{
struct
buffer buffer;
struct
mad_decoder decoder;
int
result;
/* initialize our private message structure */
buffer.start = start;
buffer.length = length;
/* configure input, output, and error functions */
mad_decoder_init(&decoder, &buffer,
input, 0
/* header */
, 0
/* filter */
, output,
error, 0
/* message */
);
/* start decoding */
result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC);
/* release the decoder */
mad_decoder_finish(&decoder);
return
result;
}
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编译： gcc -o minimad minimad.c –lmad
运行： ./minimad <test.mp3 >test.pcm
以上是将：1、标准输入重定向到MP3文件
2、标准输出重定向到解码以后的 pcm 文件
下面将pcm数据写入音频设备（即pcmplay.c程序）：
（ A.设备文件/dev/dsp
B.ioctl设置音频属性： (记得加<sys/soundcard.h>头文件）
a.采样格式
b.采样频率
c.声道
C.将pcm文件写入音频设备）
文档中pcmplay.c程序中void writefully(int fd,void *buf,int size);函数未给出，下面已补全。
pcmplay.c代码：
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#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <fcntl.h>
#include <errno.h>
#include <sys/ioctl.h>
#include <string.h>
#include <sys/soundcard.h>
void
writefully(
int
fd,
void
*buf,
int
size)
{
int
n = write(fd,buf,size);
if
(n < 0)
{
fprintf
(stderr,
"writefully error!"
,
strerror
(
errno
));
exit
(-1);
}
}
int
main(
int
argc,
char
*argv[])
{
int
handle, fd;
char
buf[1024];
if
(argc != 2)
{
fprintf
(stderr,
"usage : %s \n"
, argv[0]);
exit
(-1);
}
if
((fd = open(argv[1], O_RDONLY)) < 0)
{
fprintf
(stderr,
"Can't open sound file!\n"
);
exit
(-2);
}
if
((handle = open(
"/dev/dsp"
, O_WRONLY))<0)
{
fprintf
(stderr,
"Can't open system file /dev/dsp!\n"
);
exit
(-2);
}
#if 1
//设置声道
int
channels = 2;
int
result = ioctl(handle, SNDCTL_DSP_CHANNELS, &channels);
if
( result == -1 )
{
perror
(
"ioctl channel number"
);
return
-1;
}
//设置采样格式：AFMT_S16_LE
int
format = AFMT_S16_LE;
result = ioctl(handle, SNDCTL_DSP_SETFMT, &format);
if
( result == -1 )
{
perror
(
"ioctl sample format"
);
return
-1;
}
//设置采样频率44.1
//int rate = 22050;
int
rate = 44100;
result = ioctl(handle, SNDCTL_DSP_SPEED, &rate);
if
( result == -1 )
{
perror
(
"ioctl sample format"
);
return
-1;
}
#endif
int
n;
while
((n=read(fd,buf,
sizeof
(buf))))
{
writefully(handle,buf,n);
}
close(fd);
close(handle);
exit
(0);
}
|
编译： gcc -o pcmplay pcmplay.c
运行： ./pcmplay test.pcm
如此即可先将.mp3文件整个解压到.pcm文件中，再通过将.pcm文件写入音频设备进行.mp3音乐播放。
下面简易实现.mp3音乐文件的编解码边播放程序的编写。
------------------------------------------------------------------------------------------------—————————————————————
三：编解码边播放，用Libmad设计一个简单的MP3播放器：
“Libmad库的使用.pdf”文档中MP3player.c程序有些许缺失或错误，现已改正，程序如下：
MP3player.c
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#include "mad.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/mman.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
#define BUFSIZE 8192
/*
* This is a private message structure. A generic pointer to this structure
* is passed to each of the callback functions. Put here any data you need
* to access from within the callbacks.
*/
struct
buffer {
FILE
*fp;
/*file pointer*/
unsigned
int
flen;
/*file length*/
unsigned
int
fpos;
/*current position*/
unsigned
char
fbuf[BUFSIZE];
/*buffer*/
unsigned
int
fbsize;
/*indeed size of buffer*/
};
typedef
struct
buffer mp3_file;
int
soundfd;
/*soundcard file*/
unsigned
int
prerate = 0;
/*the pre simple rate*/
int
writedsp(
int
c)
{
return
write(soundfd, (
char
*)&c, 1);
}
void
set_dsp()
{
int
rate = 44100;
// int rate = 96000;
int
format = AFMT_S16_LE;
int
channels = 2;
int
value;
soundfd = open(
"/dev/dsp"
, O_WRONLY);
ioctl(soundfd,SNDCTL_DSP_SPEED,&rate);
ioctl(soundfd, SNDCTL_DSP_SETFMT, &format);
ioctl(soundfd, SNDCTL_DSP_CHANNELS, &channels);
/*
value = 16;
ioctl(soundfd,SNDCTL_DSP_SAMPLESIZE,&value);
value = 0;
ioctl(soundfd,SNDCTL_DSP_STEREO,&value);
*/
}
/*
* This is perhaps the simplest example use of the MAD high-level API.
* Standard input is mapped into memory via mmap(), then the high-level API
* is invoked with three callbacks: input, output, and error. The output
* callback converts MAD's high-resolution PCM samples to 16 bits, then
* writes them to standard output in little-endian, stereo-interleaved
* format.
*/
static
int
decode(mp3_file *mp3fp);
int
main(
int
argc,
char
*argv[])
{
long
flen, fsta, fend;
int
dlen;
mp3_file *mp3fp;
if
(argc != 2)
return
1;
mp3fp = (mp3_file *)
malloc
(
sizeof
(mp3_file));
if
((mp3fp->fp =
fopen
(argv[1],
"r"
)) == NULL)
{
printf
(
"can't open source file.\n"
);
return
2;
}
fsta =
ftell
(mp3fp->fp);
fseek
(mp3fp->fp, 0, SEEK_END);
fend =
ftell
(mp3fp->fp);
flen = fend - fsta;
fseek
(mp3fp->fp, 0, SEEK_SET);
fread
(mp3fp->fbuf, 1, BUFSIZE, mp3fp->fp);
mp3fp->fbsize = BUFSIZE;
mp3fp->fpos = BUFSIZE;
mp3fp->flen = flen;
set_dsp();
decode(mp3fp);
close(soundfd);
fclose
(mp3fp->fp);
return
0;
}
/*
* This is the input callback. The purpose of this callback is to (re)fill
* the stream buffer which is to be decoded. In this example, an entire file
* has been mapped into memory, so we just call mad_stream_buffer() with the
* address and length of the mapping. When this callback is called a second
* time, we are finished decoding.
*/
static
enum
mad_flow input(
void
*data,
struct
mad_stream *stream)
{
mp3_file *mp3fp;
int
ret_code;
int
unproc_data_size;
/*the unprocessed data's size*/
int
copy_size;
mp3fp = (mp3_file *)data;
if
(mp3fp->fpos <= mp3fp->flen)
{
unproc_data_size = stream->bufend - stream->next_frame;
memcpy
(mp3fp->fbuf, mp3fp->fbuf+mp3fp->fbsize-unproc_data_size, unproc_data_size);
copy_size = BUFSIZE - unproc_data_size;
if
(mp3fp->fpos + copy_size > mp3fp->flen)
{
copy_size = mp3fp->flen - mp3fp->fpos;
}
fread
(mp3fp->fbuf+unproc_data_size, 1, copy_size, mp3fp->fp);
mp3fp->fbsize = unproc_data_size + copy_size;
mp3fp->fpos += copy_size;
/*Hand off the buffer to the mp3 input stream*/
mad_stream_buffer(stream, mp3fp->fbuf, mp3fp->fbsize);
ret_code = MAD_FLOW_CONTINUE;
}
else
{
ret_code = MAD_FLOW_STOP;
}
return
ret_code;
}
/*
* The following utility routine performs simple rounding, clipping, and
* scaling of MAD's high-resolution samples down to 16 bits. It does not
* perform any dithering or noise shaping, which would be recommended to
* obtain any exceptional audio quality. It is therefore not recommended to
* use this routine if high-quality output is desired.
*/
static
inline
signed
int
scale(mad_fixed_t sample)
{
/* round */
sample += (1L <= MAD_F_FRACBITS - 16);
if
(sample >= MAD_F_ONE)
sample = MAD_F_ONE - 1;
else
if
(sample < -MAD_F_ONE)
sample = -MAD_F_ONE;
return
sample >> (MAD_F_FRACBITS + 1 - 16);
}
/*
* This is the output callback function. It is called after each frame of
* MPEG audio data has been completely decoded. The purpose of this callback
* is to output (or play) the decoded PCM audio.
*/
static
enum
mad_flow output(
void
*data,
struct
mad_header
const
*header,
struct
mad_pcm *pcm)
{
unsigned
int
nchannels, nsamples;
unsigned
int
rate;
mad_fixed_t
const
*left_ch, *right_ch;
/* pcm->samplerate contains the sampling frequency */
rate = pcm->samplerate;
nchannels = pcm->channels;
nsamples = pcm->length;
left_ch = pcm->samples[0];
right_ch = pcm->samples[1];
/* update the sample rate of dsp*/
if
(rate != prerate)
{
ioctl(soundfd, SNDCTL_DSP_SPEED, &rate);
prerate = rate;
}
while
(nsamples--)
{
signed
int
sample;
/* output sample(s) in 16-bit signed little-endian PCM */
sample = scale(*left_ch++);
writedsp((sample >> 0) & 0xff);
writedsp((sample >> 8) & 0xff);
if
(nchannels == 2)
{
sample = scale(*right_ch++);
writedsp((sample >> 0) & 0xff);
writedsp((sample >> 8) & 0xff);
}
}
return
MAD_FLOW_CONTINUE;
}
/*
* This is the error callback function. It is called whenever a decoding
* error occurs. The error is indicated by stream->error; the list of
* possible MAD_ERROR_* errors can be found in the mad.h (or stream.h)
* header file.
*/
static
enum
mad_flow error(
void
*data,
struct
mad_stream *stream,
struct
mad_frame *frame)
{
mp3_file *mp3fp = data;
fprintf
(stderr,
"decoding error 0x%04x (%s) at byte offset %u\n"
,
stream->error, mad_stream_errorstr(stream),
stream->this_frame - mp3fp->fbuf);
/* return MAD_FLOW_BREAK here to stop decoding (and propagate an error) */
return
MAD_FLOW_CONTINUE;
}
/*
* This is the function called by main() above to perform all the decoding.
* It instantiates a decoder object and configures it with the input,
* output, and error callback functions above. A single call to
* mad_decoder_run() continues until a callback function returns
* MAD_FLOW_STOP (to stop decoding) or MAD_FLOW_BREAK (to stop decoding and
* signal an error).
*/
static
int
decode(mp3_file *mp3fp)
{
struct
mad_decoder decoder;
int
result;
/* configure input, output, and error functions */
mad_decoder_init(&decoder, mp3fp,
input, 0
/* header */
, 0
/* filter */
, output,
error, 0
/* message */
);
/* start decoding */
result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC);
/* release the decoder */
mad_decoder_finish(&decoder);
return
result;
}
|
编译：gcc -o mp3player MP3player.c -lmad
运行：./mp3player xxx.mp3
至此，一个简易MP3播放器就写好了。
程序已亲自验证，请放心阅览。如有错误，欢迎批评指正。
享受阳光，享受生活。愿与大家共同进步。
本文转自 006玩命 51CTO博客，原文链接：http://blog.51cto.com/weiyuqingcheng/1362179，如需转载请自行联系原作者
