一、MP3播放器：

本设计根据libmad库中minimad.c改写成的，保留了原始的英文注释，minimad.c实现了MP3的解码成PCM音频数据，打印到屏幕上。本设计添加了alsa的播放设置函数，以及在解码output的函数中，将输出写入到声卡中，实现了MP3 文件的解码播放。

注意：本设计编译之前需要编译libmad库， 编译时需要连上 -lmad -lasound 的选项。

使用方法为在终端:./mp3-player + mp3 file.

mp3-player.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 <string.h>
#include<fcntl.h>
#include <stdlib.h>
#include <sys/ioctl.h>
#include <alsa/asoundlib.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 set_pcm();
snd_pcm_t*             handle=NULL;        //PCI设备句柄
snd_pcm_hw_params_t*   params=NULL;//硬件信息和PCM流配置

int main(int argc, char *argv[])
{
  struct stat stat;
  void *fdm;

if (argc != 2)
    {
    printf("Usage: minimad + mp3 file name");
    return 1;
    }
  int fd;
  fd=open(argv[1],O_RDWR);
  if(fd<0)
  {
    perror("open file failed:");
    return 1;
  }    
 
  if (fstat(fd, &stat) == -1 ||stat.st_size == 0)
  {
    printf("fstat failed:\n");
    return 2;
  }
   //printf("stat.st_size=%d\n",stat.st_size);
  
  fdm = mmap(0, stat.st_size, PROT_READ, MAP_SHARED, fd, 0);
  if (fdm == MAP_FAILED)
    return 3;

if(set_pcm()!=0)                 //设置pcm 参数
    {
        printf("set_pcm fialed:\n");
        return 1;   
    }
  decode(fdm, stat.st_size);

if (munmap(fdm, stat.st_size) == -1)
    return 4;

snd_pcm_drain(handle);
    snd_pcm_close(handle);

return 0;
}

int set_pcm()
{
    int    rc;     
    int  dir=0;
    int rate = 44100;;                /* 采样频率 44.1KHz*/
    int format = SND_PCM_FORMAT_S16_LE; /*     量化位数 16      */
    int channels = 2;                 /*     声道数 2           */
    
    rc=snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
        if(rc<0)
        {
                perror("\nopen PCM device failed:");
                exit(1);
        }
    snd_pcm_hw_params_alloca(&params); //分配params结构体
        
    rc=snd_pcm_hw_params_any(handle, params);//初始化params
        if(rc<0)
        {
                perror("\nsnd_pcm_hw_params_any:");
                exit(1);
        }
    rc=snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);                                 //初始化访问权限
        if(rc<0)
        {
                perror("\nsed_pcm_hw_set_access:");
                exit(1);

}
        
    rc=snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);             //设置16位采样精度  
        if(rc<0)
       {
            perror("snd_pcm_hw_params_set_format failed:");
            exit(1);
        } 
        
    rc=snd_pcm_hw_params_set_channels(handle, params, channels);  //设置声道,1表示单声>道，2表示立体声
        if(rc<0)
        {
                perror("\nsnd_pcm_hw_params_set_channels:");
                exit(1);
        }    
        
     rc=snd_pcm_hw_params_set_rate_near(handle, params, &rate, &dir);  //设置>频率
        if(rc<0)
        {
                perror("\nsnd_pcm_hw_params_set_rate_near:");
                exit(1);
        }   
        
         
    rc = snd_pcm_hw_params(handle, params);
        if(rc<0)
        {
        perror("\nsnd_pcm_hw_params: ");
        exit(1);
        } 
   
    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;

printf("this is input\n");
  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,n;
  mad_fixed_t const *left_ch, *right_ch;

/* pcm->samplerate contains the sampling frequency */

nchannels = pcm->channels;
  n=nsamples  = pcm->length;
  left_ch   = pcm->samples[0];
  right_ch  = pcm->samples[1];
  
  unsigned char Output[6912], *OutputPtr;  
  int fmt, wrote, speed, exact_rate, err, dir; 
  
  
//   printf("This is output\n");
   
   
 
   OutputPtr = Output;  
   
   while (nsamples--) 
   {
    signed int sample;

/* output sample(s) in 16-bit signed little-endian PCM */
    
    sample = scale(*left_ch++);
   
    *(OutputPtr++) = sample >> 0;  
    *(OutputPtr++) = sample >> 8;  
    if (nchannels == 2)  
        {  
            sample = scale (*right_ch++);  
            *(OutputPtr++) = sample >> 0;  
            *(OutputPtr++) = sample >> 8;  
        }  
    
  
  }
 
    OutputPtr = Output;  
    snd_pcm_writei (handle, OutputPtr, n);  
    OutputPtr = Output;

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;
  printf("this is mad_flow error\n");
  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;
}

二、WAV播放器：

本设计思路：先打开一个普通wav音频文件，从定义的文件头前面的44个字节中，取出文件头的定义消息，置于一个文件头的结构体中。然后打开alsa音频驱动，从文件头结构体取出采样精度，声道数，采样频率三个重要参数，利用alsa音频驱动的API设置好参数，最后打开wav文件，定位到数据区，把音频数据依次写到音频驱动中去，开始播放，当写入完成后，退出写入的循环。

注意：本设计需要alsa的libasound-dev的库，编译链接时需要连接 —lasound.

#include<stdio.h>
#include<stdlib.h>
#include <string.h>
#include <alsa/asoundlib.h>

struct WAV_HEADER
{
    char rld[4];    //riff 标志符号
    int rLen;   
    char wld[4];    //格式类型（wave）
    char fld[4];    //"fmt"

int fLen;   //sizeof(wave format matex)
    
    short wFormatTag;   //编码格式
    short wChannels;    //声道数
    int   nSamplesPersec ;  //采样频率
    int   nAvgBitsPerSample;//WAVE文件采样大小
    short  wBlockAlign; //块对齐
    short wBitsPerSample;   //WAVE文件采样大小
    
    char dld[4];        //”data“
    int wSampleLength;  //音频数据的大小

} wav_header;

int set_pcm_play(FILE *fp);

int main(int argc,char *argv[])
{

if(argc!=2)
    {
        printf("Usage:wav-player+wav file name\n");
        exit(1);
    }

int nread;
    FILE *fp;
    fp=fopen(argv[1],"rb");
    if(fp==NULL)
    {
        perror("open file failed:\n");
        exit(1);
    }
    
    nread=fread(&wav_header,1,sizeof(wav_header),fp);
    printf("nread=%d\n",nread);
    
    //printf("RIFF 标志%s\n",wav_header.rld);
    printf("文件大小rLen：%d\n",wav_header.rLen);
    //printf("wld=%s\n",wav_header.wld);
    //printf("fld=%s\n",wav_header.fld);
    
   // printf("fLen=%d\n",wav_header.fLen);
    
    //printf("wFormatTag=%d\n",wav_header.wFormatTag);
    printf("声道数：%d\n",wav_header.wChannels);
    printf("采样频率：%d\n",wav_header.nSamplesPersec);
    //printf("nAvgBitsPerSample=%d\n",wav_header.nAvgBitsPerSample);
    //printf("wBlockAlign=%d\n",wav_header.wBlockAlign);
    printf("采样的位数：%d\n",wav_header.wBitsPerSample);
    
   // printf("data=%s\n",wav_header.dld);
    printf("wSampleLength=%d\n",wav_header.wSampleLength);
    
    
    
    
    
    set_pcm_play(fp);
    return 0;
}

int set_pcm_play(FILE *fp)
{
        int    rc;
        int    ret;
        int    size;
        snd_pcm_t*       handle;        //PCI设备句柄
        snd_pcm_hw_params_t*      params;//硬件信息和PCM流配置
        unsigned int       val;
        int                dir=0;
        snd_pcm_uframes_t  frames;
        char   *buffer;
        int channels=wav_header.wChannels;
        int frequency=wav_header.nSamplesPersec;
        int bit=wav_header.wBitsPerSample;
        int datablock=wav_header.wBlockAlign;
        unsigned char ch[100];  //用来存储wav文件的头信息
    
    
        
        rc=snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
        if(rc<0)
        {
                perror("\nopen PCM device failed:");
                exit(1);
        }

snd_pcm_hw_params_alloca(&params); //分配params结构体
        if(rc<0)
        {
                perror("\nsnd_pcm_hw_params_alloca:");
                exit(1);
        }
         rc=snd_pcm_hw_params_any(handle, params);//初始化params
        if(rc<0)
        {
                perror("\nsnd_pcm_hw_params_any:");
                exit(1);
        }
        rc=snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);                                 //初始化访问权限
        if(rc<0)
        {
                perror("\nsed_pcm_hw_set_access:");
                exit(1);

}

//采样位数
        switch(bit/8)
        {
        case 1:snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_U8);
                break ;
        case 2:snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
                break ;
        case 3:snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S24_LE);
                break ;

}
        rc=snd_pcm_hw_params_set_channels(handle, params, channels);  //设置声道,1表示单声>道，2表示立体声
        if(rc<0)
        {
                perror("\nsnd_pcm_hw_params_set_channels:");
                exit(1);
        }
        val = frequency;
        rc=snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);  //设置>频率
        if(rc<0)
        {
                perror("\nsnd_pcm_hw_params_set_rate_near:");
                exit(1);
        }

rc = snd_pcm_hw_params(handle, params);
        if(rc<0)
        {
        perror("\nsnd_pcm_hw_params: ");
        exit(1);
        }

rc=snd_pcm_hw_params_get_period_size(params, &frames, &dir);  /*获取周期
长度*/
        if(rc<0)
        {
                perror("\nsnd_pcm_hw_params_get_period_size:");
                exit(1);
        }

size = frames * datablock;   /*4 代表数据快长度*/

buffer =(char*)malloc(size);
    fseek(fp,58,SEEK_SET);  //定位歌曲到数据区

while (1)
        {
                memset(buffer,0,sizeof(buffer));
                ret = fread(buffer, 1, size, fp);
                if(ret == 0)
                {
                        printf("歌曲写入结束\n");
                        break;
                }
                 else if (ret != size)
                {
                 }
                // 写音频数据到PCM设备  
        while(ret = snd_pcm_writei(handle, buffer, frames)<0)
           {
                 usleep(2000);  
                 if (ret == -EPIPE)
                {
                  /* EPIPE means underrun */
                  fprintf(stderr, "underrun occurred\n");
                  //完成硬件参数设置，使设备准备好  
                  snd_pcm_prepare(handle);
                 }
                 else if (ret < 0)
                 {
                          fprintf(stderr,
                      "error from writei: %s\n",
                      snd_strerror(ret));
                 }
            }

}

snd_pcm_drain(handle);
        snd_pcm_close(handle);
        free(buffer);
        return 0;
}