linux 音频编程

http://blog.csdn.net/sea918/article/details/7249216

 

1、音频开发模型：

OSS(open sound system)  linux/unix 平台的上早期的统一音频接口。linux kernl 2.6 版本以前其它提供两种设备文件以供编程。 常用的操作函数为open、close、read、write、ioctl.

(/dev/dsp录音设备文件/dev/audio播放设备文件)

ALSA（a）目前流行的编译框架。linux 2.6 版本发后支持。

提供统一的编程接口：snd_pcm_open、snd_pcm_close、snd_pcm_hw_params

基设备文件为:/dev/snd/pcmC0D0p    /dev/snd/pcmC0D0c  /dev/snd/pcmC0D1p   /dev/snd/timer

可以通过bash命令查看 alsa 驱动版本：

root@ubuntu:cat  /proc/asound/version

Advanced linux Sound Architecture Driver Version 1.0.23

2,Alsa-lib 编译

a,下载并安装 alsa-lib库

root@ubuntu: tar -xvf  alsa-lib-1.0.13.tar.bz2

root@ubuntu:./configure

root@ubuntu:make

root@ubuntu: make install

3，编程

a,添加头文件

#include

b,编程录音代码.

*/  
  
/* Use the newer ALSA API */  
#define ALSA_PCM_NEW_HW_PARAMS_API  
#include   
#include   
#include   
  
#define LENGTH    3   //录音时间,秒  
#define RATE    9600 //采样频率  
#define SIZE    16   //量化位数  
#define CHANNELS 1   //声道数目  
#define RSIZE    8    //buf的大小，  
  
/********以下是wave格式文件的文件头格式说明******/  
/*------------------------------------------------ 
|             RIFF WAVE Chunk                  | 
|             ID = 'RIFF'                     | 
|             RiffType = 'WAVE'                | 
------------------------------------------------ 
|             Format Chunk                     | 
|             ID = 'fmt '                      | 
------------------------------------------------ 
|             Fact Chunk(optional)             | 
|             ID = 'fact'                      | 
------------------------------------------------ 
|             Data Chunk                       | 
|             ID = 'data'                      | 
------------------------------------------------*/  
/**********以上是wave文件格式头格式说明***********/  
/*wave 文件一共有四个Chunk组成，其中第三个Chunk可以省略，每个Chunk有标示（ID）, 
大小（size,就是本Chunk的内容部分长度）,内容三部分组成*/  
struct fhead  
{  
/****RIFF WAVE CHUNK*/  
unsigned char a[4];//四个字节存放'R','I','F','F'  
long int b;        //整个文件的长度-8;每个Chunk的size字段，都是表示除了本Chunk的ID和SIZE字段外的长度;  
unsigned char c[4];//四个字节存放'W','A','V','E'  
/****RIFF WAVE CHUNK*/  
/****Format CHUNK*/  
unsigned char d[4];//四个字节存放'f','m','t',''  
long int e;       //16后没有附加消息，18后有附加消息；一般为16，其他格式转来的话为18  
short int f;       //编码方式，一般为0x0001;  
short int g;       //声道数目，1单声道，2双声道;  
long int h;        //采样频率;  
long int i;        //每秒所需字节数;  
short int j;       //每个采样需要多少字节，若声道是双，则两个一起考虑;  
short int k;       //即量化位数  
/****Format CHUNK*/  
/***Data Chunk**/  
   unsigned char p[4];//四个字节存放'd','a','t','a'  
long int q;        //语音数据部分长度，不包括文件头的任何部分  
}wavehead;//定义WAVE文件的文件头结构体  
  
  
int startRecord(void)  
{  
long loops;  
int rc;  
int size;  
snd_pcm_t *handle;  
snd_pcm_hw_params_t *params;  
unsigned int val;  
int dir;  
snd_pcm_uframes_t frames;  
char *buffer;  
int fd_f;  
int status;  
  
  
  
/*以下wave 文件头赋值*/  
wavehead.a[0]='R';  
wavehead.a[1]='I';  
wavehead.a[2]='F';  
wavehead.a[3]='F';  
wavehead.b=LENGTH*RATE*CHANNELS*SIZE/8-8;  
wavehead.c[0]='W';  
wavehead.c[1]='A';  
wavehead.c[2]='V';  
wavehead.c[3]='E';  
wavehead.d[0]='f';  
wavehead.d[1]='m';  
wavehead.d[2]='t';  
wavehead.d[3]=' ';  
wavehead.e=16;  
wavehead.f=1;  
wavehead.g=CHANNELS;  
wavehead.h=RATE;  
wavehead.i=RATE*CHANNELS*SIZE/8;  
wavehead.j=CHANNELS*SIZE/8;  
wavehead.k=SIZE;  
wavehead.p[0]='d';  
wavehead.p[1]='a';  
wavehead.p[2]='t';  
wavehead.p[3]='a';  
wavehead.q=LENGTH*RATE*CHANNELS*SIZE/8;  
/*以上wave 文件头赋值*/  
  
  
/* Open PCM device for recording (capture). */  
rc = snd_pcm_open(&handle, "default",  
                    SND_PCM_STREAM_CAPTURE, 0);  
if (rc < 0) {  
    fprintf(stderr,  
            "unable to open pcm device: %s\n",  
            snd_strerror(rc));  
    exit(1);  
}  
  
/* Allocate a hardware parameters object. */  
snd_pcm_hw_params_alloca(¶ms);  
  
/* Fill it in with default values. */  
snd_pcm_hw_params_any(handle, params);  
  
/* Set the desired hardware parameters. */  
  
/* Interleaved mode */  
snd_pcm_hw_params_set_access(handle, params,  
                      SND_PCM_ACCESS_RW_INTERLEAVED);  
  
/* Signed 16-bit little-endian format */  
snd_pcm_hw_params_set_format(handle, params,  
                              SND_PCM_FORMAT_S16_LE);  
  
/* Two channels (stereo) */  
snd_pcm_hw_params_set_channels(handle, params, CHANNELS);  
  
/* 44100 bits/second sampling rate (CD quality) */  
val = RATE;  
snd_pcm_hw_params_set_rate_near(handle, params,  
                                  &val, &dir);  
  
/* Set period size to 32 frames. */  
frames = 32;  
snd_pcm_hw_params_set_period_size_near(handle,  
                              params, &frames, &dir);  
  
/* Write the parameters to the driver */  
rc = snd_pcm_hw_params(handle, params);  
if (rc < 0) {  
    fprintf(stderr,  
            "unable to set hw parameters: %s\n",  
            snd_strerror(rc));  
    exit(1);  
}  
  
/* Use a buffer large enough to hold one period */  
snd_pcm_hw_params_get_period_size(params,  
                                      &frames, &dir);  
size = frames * 2; /* 2 bytes/sample, 2 channels */  
buffer = (char *) malloc(size);  
  
/* We want to loop for 5 seconds */  
snd_pcm_hw_params_get_period_time(params,  
                                         &val, &dir);  
loops = 5000000 / val;  
  
  
  
  
if(( fd_f = open("./sound.wav", O_CREAT|O_RDWR,0777))==-1)//创建一个wave格式语音文件  
{  
    perror("cannot creat the sound file");  
}  
if((status = write(fd_f, &wavehead, sizeof(wavehead)))==-1)//写入wave文件的文件头  
{  
   perror("write to sound'head wrong!!");  
}  
  
while (loops > 0) {  
    loops--;  
    rc = snd_pcm_readi(handle, buffer, frames);  
    if (rc == -EPIPE) {  
      /* EPIPE means overrun */  
      fprintf(stderr, "overrun occurred\n");  
      snd_pcm_prepare(handle);  
    } else if (rc < 0) {  
      fprintf(stderr,  
              "error from read: %s\n",  
              snd_strerror(rc));  
    } else if (rc != (int)frames) {  
      fprintf(stderr, "short read, read %d frames\n", rc);  
    }  
  
    if(write(fd_f, buffer, size)==-1)  
    {  
         perror("write to sound wrong!!");  
    }  
    if (rc != size)  
      fprintf(stderr,  
              "short write: wrote %d bytes\n", rc);  
}  
  
snd_pcm_drain(handle);  
snd_pcm_close(handle);  
free(buffer);  
close(fd_f);  
  
return 0;  
}

 

OSS的编程

 

Linux下和声卡相关的文件有许多，如采集数字样本的/dev/dsp文件，针对混音器的/dev/mixer文件以及用于音序器的/dev /sequencer等。文件/dev/audio是一个基于兼容性考虑的声音设备文件，它实际是到上述数字设备的一个映射，它最大的特色或许是对诸如 wav这类文件格式的直接支持。我们下面的例子即使用了此设备文件实现了一个简单的录音机：我们从声卡设备（当然要用麦克风）读取音频数据，并将它存放到 文件test.wav中去。要播放这个wav文件，只要如前面所述，使用命令cat test.wav >/dev/audio即可，当然你也可以用Linux下其他的多媒体软件来播放这个文件。

 

/* 此文件中定义了下面所有形如SND_的变量*/
#include 
#include 
#include 
#include 
#include 
#include 
#include 
#include <string.h>
#include 
 
#define OPEN_DSP_FAILED     0x00000001      /*打开  dsp 失败!*/
#define SAMPLERATE_STATUS     0x00000002    /*samplerate status failed*/
#define SET_SAMPLERATE_FAILED  0x00000003   /*set samplerate failed*/
#define CHANNELS_STATUS       0x00000004    /*Channels status failed*/
#define SET_CHANNELS_FAILED    0x00000005   /*set channels failed*/
#define FMT_STATUS       0x00000006        /*FMT status failed*/
#define SET_FMT_FAILED     0x00000007       /*set fmt failed*/
#define OPEN_FILE_FAILED        0x00000008    /*opem filed failed*/
 
int P8100_Audio_Play(char *pathname,int nSampleRate,int nChannels,int fmt)
{
int dsp_fd,mix_fd,status,arg;
dsp_fd = open("/dev/dsp" , O_RDWR);   /*open dsp*/
if(dsp_fd < 0)
{
  return  OPEN_DSP_FAILED;
}
arg = nSampleRate;
status = ioctl(dsp_fd,SOUND_PCM_WRITE_RATE,&arg); /*set samplerate*/
if(status < 0)
{
  close(dsp_fd);
  return SAMPLERATE_STATUS;
}
if(arg != nSampleRate)
{
  close(dsp_fd);
  return SET_SAMPLERATE_FAILED;
}
arg = nChannels;  /*set channels*/  
status = ioctl(dsp_fd, SOUND_PCM_WRITE_CHANNELS, &arg);
if(status < 0)
{
  close(dsp_fd);
  return CHANNELS_STATUS;
}
if( arg != nChannels)
{
  close(dsp_fd);
  return SET_CHANNELS_FAILED;
}
arg = fmt; /*set bit fmt*/
status = ioctl(dsp_fd, SOUND_PCM_WRITE_BITS, &arg);
if(status < 0)
{
  close(dsp_fd);
  return FMT_STATUS;
}
if(arg != fmt)
{
  close(dsp_fd);
  return SET_FMT_FAILED;
}/*到此设置好了DSP的各个参数*/           
FILE *file_fd = fopen(pathname,"r");
if(file_fd == NULL)
{
  close(dsp_fd);
  return OPEN_FILE_FAILED;
}
int num = 3*nChannels*nSampleRate*fmt/8;
int get_num;
char buf[num];
while(feof(file_fd) == 0)
{
  get_num = fread(buf,1,num,file_fd);
  write(dsp_fd,buf,get_num);
  if(get_num != num)
  {
   close(dsp_fd);
   fclose(file_fd);
   return 0;
  }
}
close(dsp_fd);
fclose(file_fd);
return 0;
}
  
int main()
{
    int value;
 
    value = P8100_Audio_Play("/windows/C/WINDOWS/Media/Windows Startup.wav",44100,2,16);
    //注意播放文件的路径哦！！
    fprintf(stderr,"value is %d",value);
    return 0;
}