一.编译libmad
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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
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3.make 编译 并记录信息
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Tips
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改动makefile ,删除 " --fforce-mem "
4. make install 安装 并记录信息
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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