libmad学习进阶3-----基于oss音频驱动架构的一个mp3播放器

/*
modify by hfl 2014-2-16
*/
# include <stdio.h>
# include <unistd.h>
# include <sys/stat.h>
# include <sys/mman.h>


# include "mad.h"
#include<sys/types.h>
#include<sys/stat.h>
#include<fcntl.h>
#include<stdlib.h>




#include <sys/ioctl.h>
#include <sys/soundcard.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.
 */
 //#define printf    
static Get_file_length(char *PATH);
static int init_dsp();
static int decode(unsigned char const *, unsigned long);
static enum mad_flow outputplay(void *data,
		     struct mad_header const *header,
		     struct mad_pcm *pcm);
int main(int argc, char *argv[])
{
printf("The main is start!\n");
  struct stat stat;
  void *fdm;


  int  fd;
  //char buffer1[80000];
  printf("###The input file is %s  ! the arc=%d###\n",argv[1],argc);
  if (argc == 1)
   {
   printf("The argc is wrong!\n");
    return 1;
 }
  
#if 0
  if (fstat(STDIN_FILENO, &stat) == -1 ||
      stat.st_size == 0)
    return 2;
#endif


 fd =open(argv[1],O_RDWR);
if(-1==fd)
{
   printf("sorry,The file open is faild!\n");
}
else
{
 printf("The file open is sucessed!\n");
}
//read(fd,buffer1,sizeof(buffer1));
//printf("%s", buffer1);
stat.st_size = Get_file_length(argv[1]);
printf("The file size is %d\n",stat.st_size );
printf("The Map is begin ok!\n");


fdm = mmap(0, stat.st_size, PROT_READ, MAP_SHARED, fd, 0);
  if (fdm == MAP_FAILED)
  	{
  	printf("mmap is failed\n");
    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;
};
int id;
int flag=0;
/*初始化音频设备*/
int init_dsp(int rate,int channels)
{
        
        /* 采样频率 44.1KHz*/
	int format; /* 量化位数 16*/
	 /* 声道数 2*/
	int setting; /* 高16位标明分片最大序号,低16位标明缓冲区的尺寸*/
	if ( ( id = open ( "/dev/dsp", O_WRONLY) ) < 0 )
	{
		fprintf (stderr, " Can't open sound device!\n");
		exit ( -1 ) ;
	}
	/* 此项在Virtual PC中可减少电流声 */
	setting = 0x0002000F;
	ioctl(id, SNDCTL_DSP_SETFRAGMENT, &setting);


	//rate = 44100;
	ioctl(id, SNDCTL_DSP_SPEED, &rate);


	format = AFMT_S16_LE;
	ioctl(id, SNDCTL_DSP_SETFMT, &format);


	//channels = 2;
	ioctl(id, SNDCTL_DSP_CHANNELS, &channels);
	printf( "The Dsp init is oss ok!\n");
	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)
{
  struct buffer *buffer = data;


  if (!buffer->length)
    return MAD_FLOW_STOP;


  mad_stream_buffer(stream, buffer->start, buffer->length);


  buffer->length = 0;
printf("1111");
  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);
}


static int Get_file_length(char *PATH)
{  
     FILE *fp;
     fp=fopen(PATH,"r");
    if(!fp)
   {
   printf("sorry,The file open is faild!\n");
   }
   else
    {
    printf("The file open is sucessed!\n");
   }
   fseek(fp, 0L,SEEK_END);
   return (ftell(fp));
}
/*
 * 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;
  static FILE *fdout;
  char buf[1];
  
  /* pcm->samplerate contains the sampling frequency */
 fdout= fopen("mypcm.pcm","ab+");
  if(!fdout)
  	{
  	  printf("open is failed\n");
  	}
  else
  	printf("out open is ok\n");
  nchannels = pcm->channels;
  nsamples  = pcm->length;
  left_ch   = pcm->samples[0];
  right_ch  = pcm->samples[1];
//printf("channels=%d, samples=%d\n", nchannels,nsamples);
  while (nsamples--) {
    signed int sample;


    /* output sample(s) in 16-bit signed little-endian PCM */
  // printf("2222\n");
    sample = scale(*left_ch++);
   
    buf[0]=(sample >> 0) & 0xff;
    printf("%d\t",buf[0]);
    fwrite(buf,1,1,fdout);
    
    buf[0]=(sample >> 8) & 0xff;
	   printf("%d\t",buf[0]);
    fwrite(buf,1,1,fdout);
    if (nchannels == 2) {
     sample = scale(*right_ch++);
	
     buf[0]=(sample >> 0) & 0xff;
	fwrite(buf,1,1,fdout);
 
     buf[0]=(sample >> 8) & 0xff;
	fwrite(buf,1,1,fdout);
    }
  }
fclose(fdout);
  return MAD_FLOW_CONTINUE;
}
static
enum mad_flow outputplay(void *data,
		     struct mad_header const *header,
		     struct mad_pcm *pcm)
{
  unsigned int nchannels;
  long int nsamples,samplerate;
  mad_fixed_t const *left_ch, *right_ch;
 
  char buf[1];
 
  /* pcm->samplerate contains the sampling frequency */


  nchannels = pcm->channels;
  nsamples  = pcm->length;/* 这个不是采样位，而一帧的数据长度12*3(采样)*32(子带)=1152*/
  left_ch   = pcm->samples[0];
  right_ch  = pcm->samples[1];
  samplerate=pcm->samplerate;
  


  if(!flag)
  	{
  printf("channels=%d, samples2=%ld,flag=%d\n", nchannels,samplerate,flag);
		printf("init dsp is begin\n");
  init_dsp(samplerate,nchannels);
   flag++;
  	}
  while (nsamples--) {
    signed int sample;


    /* output sample(s) in 16-bit signed little-endian PCM */
  // printf("2222\n");
    sample = scale(*left_ch++);
   
    buf[0]=(sample >> 0) & 0xff;
    //printf("%d\t",buf[0]);
     write(id,buf,1);
    buf[0]=(sample >> 8) & 0xff;
	  // printf("%d\t",buf[0]);
  write(id,buf,1);
    if (nchannels == 2) {
     sample = scale(*right_ch++);
	
     buf[0]=(sample >> 0) & 0xff;
	write(id,buf,1);
 
     buf[0]=(sample >> 8) & 0xff;
	write(id,buf,1);
    }
  }
//fclose(fdout);
  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 */, outputplay,
		   error, 0 /* message */);


  /* start decoding */


  result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC);


  /* release the decoder */


  mad_decoder_finish(&decoder);


  return result;
}

以上就是通过libmad将mp3先解码成pcm,然后将pcm直接扔到/ dev/dsp音频设备中，但dsp音频设备属于oss架构，已经逐渐被alsa驱动取代，后续会介绍基于alsa驱动架构的mp3播放器