Linux下基于Libmad库的MP3音乐播放器编写
linux下基于Libmad库的MP3音乐播放器编写
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
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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
/*
* 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;
}
编译： gcc -o minimad minimad.c �Clmad
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：
#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。
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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
#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。
--刀刀