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
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\n");*(outputptr++)=ldata;*(outputptr++)=rdata;//printf("buflen%d\n",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",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\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;
}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\n");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\n");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\n");return -1;}while((fd_sound = open("/dev/dsp",O_WRONLY)) == -1) {printf("Can not open /dev/dsp\n");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\n");goto exit;}}else{goto exit;}//printf("sample:%d,channel:%d,bits:%d,vol_val:%d\n", 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\n");}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