Linux下基于Libmad库的MP3音乐播放器编写

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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（金山快盘附件：Libmad库的使用.zip（721.07KB）（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 –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：



#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



#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;
}