MP3/WAV 播放

一.编译libmad 



1.先下载压缩包到本地,并解压

 tar -xvzf  libmad-0.15.1b.tar.gz   -C   ./

2.进入源代码文件夹并配置

编写一个配置文件,便于< 改动和编译 >  文件内容例如以下

./configure CC=arm-linux-gcc  --host=arm-linux  --build=i686-pc-linux-gnu  --enable-fpm=arm  --enable-shared --disable-debugging --prefix=/home/tang/WIFI-Music/MPlayer/libmad-0.15.1b_install  

运行配置 并记录信息



3.make 编译 并记录信息



Tips 



改动makefile ,删除 " --fforce-mem "

4.  make  install 安装 并记录信息



tips

须要调用的库和文件为：libmad.so   mad.h 

二.编写 程序代码

1.可播放wav、mp3 两种格式代码。

< play-wav-or-mp3.c >

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <linux/types.h>
#include <fcntl.h>
#include <sys/types.h>
#include <semaphore.h>
#include <sys/stat.h> 
#include <string.h>
#include <errno.h>
#include <linux/soundcard.h>

#include <termio.h>
#include <getopt.h>
#include <time.h>
#include <strings.h>
#include <signal.h>
#include "wav.h"
#include "mad.h"
#include <sys/mman.h>


#define SND_OUT_BUF_SIZE	0x2000	

struct buffer {
  unsigned char const *start;
  unsigned long length;
};

int fd_sound;
int n;
int vol_val;
int i=0;
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;
}

static 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
enum mad_flow output(void *data,
		     struct mad_header const *header,
		     struct mad_pcm *pcm)
{
  unsigned short nchannels ,nsamples;
  unsigned int  nsamplerate;
  unsigned char ldata,rdata;
  unsigned char outputbuf[8196],*outputptr;
  int write_num;
  mad_fixed_t const *left_ch, *right_ch;

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

  nchannels = pcm->channels;
  nsamplerate = pcm->samplerate;
  n=nsamples  = pcm->length;
  left_ch   = pcm->samples[0];
  right_ch  = pcm->samples[1];
  
  if(i==0){
		int bits_set=16;
		ioctl(fd_sound, SNDCTL_DSP_SYNC, &nsamplerate);
		ioctl(fd_sound, SOUND_PCM_WRITE_RATE, &nsamplerate);
		ioctl(fd_sound, SNDCTL_DSP_SETFMT, &bits_set);
		ioctl(fd_sound, SOUND_PCM_WRITE_CHANNELS, &nchannels);
		ioctl(fd_sound, SOUND_MIXER_WRITE_VOLUME, &vol_val);
	}
	i++;
  outputptr=outputbuf;
  while (nsamples--) {
    signed int sample;

    /* output sample(s) in 16-bit signed little-endian PCM */
	sample = scale(*left_ch++);
	ldata = (sample >> 0);
	rdata = (sample >> 8);
	//printf("ssss
");
	*(outputptr++)=ldata;
	*(outputptr++)=rdata;
	//printf("buflen%d
",strlen(outputbuf[i]));

	if (nchannels == 2) 
	{
		sample = scale(*right_ch++);
		ldata = (sample >> 0);
		rdata = (sample >> 8);
		*(outputptr++)=ldata;
		*(outputptr++)=rdata;
	}
  }
	n*=4;
	outputptr=outputbuf;
	while(n)
	{
		write_num=write(fd_sound,outputptr,n);
		outputptr+=write_num;
		n-=write_num;
  		//printf("n:%d
",n);
	}
	outputptr=outputbuf;
	
  return MAD_FLOW_CONTINUE;
}


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
",
	  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;
}

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;
}


int main(int argc, char **argv)
{
	if(argc < 3)
	{
		printf("argc error
");
		return -1;
	}	
	int rate_set, bits_set, ch_set,fd_file_path=0,DataLen;
	char file_path[256]={0};
	int *psound_data_buf=NULL;
	struct stat stat;
	void *fdm;
	//ch_set=2;
	//bits_set=16;
	//rate_set=44100;
	if(sscanf(argv[1], "%s", &file_path)!= 1 ||sscanf(argv[2], "%d", &vol_val)!= 1)
	{
		printf("argv error
");
		return -1;
	}
	if(vol_val<0)
		vol_val=26;
	if(strcmp(&file_path[strlen(file_path)-4],".wav")!=0 && strcmp(&file_path[strlen(file_path)-4],".mp3")!=0)
	{
		printf("file is not wav or mp3 farmat
");
		return -1;
		
	}
	while((fd_sound = open("/dev/dsp",O_WRONLY)) == -1) {
		 printf("Can not open /dev/dsp
");
		 return -1;
	 }
	fd_file_path = open(file_path,O_RDONLY);
	if(fd_file_path == -1)
	{
		printf("fd_file_path open file error");
		goto exit;
	}

		
	if(strcmp(&file_path[strlen(file_path)-4],".wav")==0)
	{
		wav_struct FileWav;
		psound_data_buf=(int *)malloc(SND_OUT_BUF_SIZE);
		if(psound_data_buf == NULL)
			goto exit;

		memset(&FileWav,0,sizeof(FileWav));
		if((DataLen = read(fd_file_path, &FileWav, sizeof(FileWav)))>0)
		{
			if((strncmp(FileWav.rif_info.riff,RIFF_FIELD,strlen(RIFF_FIELD)) == 0)&&(strncmp(FileWav.rif_info.wave,WAVE_FIELD,strlen(WAVE_FIELD)) == 0))
			{
				rate_set=FileWav.fmt_info.sample_rate;
				ch_set=FileWav.fmt_info.channel_nb;
				bits_set=FileWav.fmt_info.bits_per_sample;
			}
			else
			{
				printf("wav head error
");
				goto exit;
			}
		}
		else
		{
			goto exit;
		}
		//printf("sample:%d,channel:%d,bits:%d,vol_val:%d
", rate_set,ch_set,bits_set,vol_val);
		ioctl(fd_sound, SNDCTL_DSP_SYNC, &rate_set);
		ioctl(fd_sound, SOUND_PCM_WRITE_RATE, &rate_set);
		ioctl(fd_sound, SNDCTL_DSP_SETFMT, &bits_set);
		ioctl(fd_sound, SOUND_PCM_WRITE_CHANNELS, &ch_set);
		ioctl(fd_sound, SOUND_MIXER_WRITE_VOLUME, &vol_val);
		while((DataLen=read(fd_file_path, psound_data_buf ,SND_OUT_BUF_SIZE))>0)
			write(fd_sound, psound_data_buf, DataLen);
			free(psound_data_buf);
	}
	/*mp3 play*/
	else if(strcmp(&file_path[strlen(file_path)-4],".mp3")==0)
	{
		if(fstat(fd_file_path,&stat)==-1||stat.st_size==0)
			goto exit;
		fdm=mmap(0,stat.st_size,PROT_READ,MAP_SHARED,fd_file_path,0);
		if(fdm==MAP_FAILED)
			goto exit;
		decode(fdm,stat.st_size);
		
	}
					

exit:	
	if(munmap(fdm,stat.st_size)==-1)
	{
		printf("munmap error
");
	}
	if(fd_file_path>0)
	{
		close(fd_file_path);
		fd_file_path=0;
	}
	if(fd_sound>0)
	{
		close(fd_sound);
		fd_sound=0;
	}
	
return 0;
}

< wav.h >

#ifndef _WAV_H_
#define _WAV_H_

/*_____ I N C L U D E S ____________________________________________________*/


/*_____ M A C R O S ________________________________________________________*/

#define WAV_HEADER_SIZE   sizeof(wav_struct)

/* RIFF info */
#define RIFF_FIELD        "RIFF"
#define WAVE_FIELD        "WAVE"

/* FMT info */
#define FMT_FIELD         "FMT "
#define FMT_LENGTH        ((unsigned long)(16))  /* data start beg of sector */
/* wave format */
#define PCM_FMT           ((unsigned short)0x0100)
/* channel number */
#define MONO              ((unsigned short)0x0100)
#define STEREO            ((unsigned short)0x0200)
/* bytes per sample */
#define ONE_BYTE          ((unsigned short)0x0100)
#define TWO_BYTE          ((unsigned short)0x0200)
/* bits per sample */
#define EIGHT_BIT         ((unsigned short)0x0800)
#define SIXTEEN_BIT       ((unsigned short)0x1000)
/* DATA info */
#define DATA_FIELD        'data'


/*_____ D E F I N I T I O N ________________________________________________*/

/* WAV Format Structure */

typedef struct
{ /* RIFF info */
  char    riff[4];
  unsigned long  pack_length;
  char    wave[4];
} riff_struct;

typedef struct
{ /* FMT info */
  char    fmt[4];
  unsigned long  fmt_length;
  unsigned short  wav_format; 
  unsigned short  channel_nb;
  unsigned long  sample_rate;
  unsigned long  bytes_per_second;
  unsigned short  bytes_per_sample;
  unsigned short  bits_per_sample;
} fmt_struct;

typedef struct
{ /* DATA info */
  char    dat[4];
  unsigned long  data_length;
} data_struct;

typedef struct
{
  riff_struct   rif_info;
  fmt_struct    fmt_info;
  data_struct   dat_info;
} wav_struct;


/*_____ D E C L A R A T I O N ______________________________________________*/



#endif  /* _WAV_H_ */
    

动态编译  arm-none-linux-gnueabi-gcc play-wav-or-mp3.c  -o  play-wav-or-mp3  -lmad  -L./ 

执行  ./play-wav-or-mp3  xxx.mp3/wav  70   

argc[1] 为播放歌曲，argc[2] 为音量大小


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原文地址：https://www.cnblogs.com/hrhguanli/p/3866542.html